RE: [Asterisk-Users] Asterisk@Home Installation Problems
David If your machine has been used for Windows, book from a DOS floppy and use FDISK to remove the partitions and try again. I've never tried to install on a machine with existing partitions but never had a failure on a machine without them. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Fulton-Howard Sent: March 10, 2005 5:09 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] [EMAIL PROTECTED] Installation Problems I just found out about Asterisk (yes, from the Slashdot posting), and I would like to set up my old computer as a dedicated box for my house using [EMAIL PROTECTED] However, when I try to install from the bootable CD, it gets to 54% of copying the image to the hard drive and then says that an error occurred because I have run out of disk space. The machine has a 10.2 GB hard drive, so I don't think space should be a problem since the image is being copied over from a CD... right? If not, what else could be the problem? Also, I noticed that when I boot with an XP CD to look at the partitions, the first one is about 800 MB, the scond one is 9 GB, and the third one is a couple hundred MB. I would assume it's trying to use the 9 GB one and the 800 MB one is the swap, right? If not, is there any command-line parameter I could use at the beginning of setup to fix things? If it helps, my specs are as follows: 450 MHz PII 256 MB SDRAM i440BX-based motherboard nVidia TNT AGP video card 3Com 3C905TX NIC Thanks, David Fulton-Howard ___ 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] OT: Best DB
On Wed, 9 Mar 2005, [EMAIL PROTECTED] wrote: For some reason I didn't think PostgreSQL was for mission critical apps. I don't think I have any reasoning behind it, just didn't think it was hardcore...sounds like i might be wrong...i'll have to look into it more. For your app, probably either MySQL or PostreSQL will work. I'm a happy MySQL user ... others are just as happy with PostgreSQL. I think it's almost what you're familiar with at this point. The differences between the two are getting smaller. MySQL traditionally was considered a very high speed database server lacking some advanced features such as transactions and triggers and some query types. Postgres was considered a slower, feature complete SQL implementation. Today, MySQL has more features that it lacked earlier - i.e. it's got transactions and additional queries, and so on. I understand that PostgreSql has also gotten faster than it used to be. So, at this point it's almost devolved into a holy war as opposed to there being any real difference. Personally I use MySQL because I find it easier to admin and configure on my FreeBSD systems than PostgreSQL, which I tend to have ongoing problems with in the spots I have to run it. I don't miss the couple of PostgreSQL features that mysql still doesn't have (but will in the near future). I'd really recommend that you look at developing the app so it is database independent - at least between MySQL and PostgreSQL. That way, you can swap from one to the other if you decide you don't like the one you pick initially. -forrest ___ 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: Zap channels intermittently bridging with SNOM190
Hi guys/girls, We are running a TDM04B card with Asterisk in a Linux box that has 15 GrandStream102 extensions and 1 SNOM190 phone which we are using as an operator console. The FXO ports in the TDM04B are plugged directly into our telecoms provider's analogue lines. Something I've picked up with the SNOM is that sometimes when there are two active incoming calls viaZap channelsthey end up being transferred to eachother. In other words two in bound callers end up being connected to eachother. I've checked with the operator and she's said that she's been disconnecting any 'idle' calls i.e. when the remote user hangs up but yet the problem still occurs every now and again. This is what I end up with when I run a 'show channels': Channel (Context Extension Pri ) State Appl. Data Zap/1-1 (default 1 ) Up Bridged Call Zap/2-1 Zap/2-1 (default 2009 1 ) Up Dial SIP/switchboard|30|tr For reference call transferring on the SNOM is being done via the 'consultation transfer' method as set out in the SNOM manual. Perhaps there is a way in Asterisk or the SNOM phone to prevent/disallow bridging of specific Zap channels ? Has anyone else come across this phenomenon before ? Thanks in advance Kindest regardsDavid Wilson___D c D a t aTel +27 33 342 7003Fax +27 33 345 4155Cell +27 82 4147413http://www.dcdata.co.za[EMAIL PROTECTED]Powered by Linux, driven by passion ! ___ "Computers are not intelligent. They only think they are." ___ 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] Single port S0 ISDN card to use in Greece
On Thu, 10 Mar 2005, Loucas Gatzoulis wrote: I'm trying to build a PBX using Asterisk. I have a single BRI ISDN line and I need to connect 4 internal normal phones and a couple of softphones on PC. I have bought a single port Billion S0 card and a TDM400 with 4 FXS modules for the intenal phones. ISDN lines here in Greece terminate to an NT1 terminal called NetMod (http://www.intracom.gr/en/products/terminal_equip/isdn_netcon_netmod.htm). My question is should I operate the billion card in TE or NT mode? To connect to the phone network it should operate in TE mode. TE is short for Terminal Equipment i.e. anything you connect to the pstn. Peter ___ 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: Zap channels intermittently bridging with SNOM190
Hi David On Thu, 10 Mar 2005, quoth David Wilson: Something I've picked up with the SNOM is that sometimes when there are two active incoming calls via Zap channels they end up being transferred to eachother. In other words two in bound callers end up being connected to eachother. I've come across this exact problem. Whenever there are two calls on hold and you push the transfer button, it transfers them both together. The solution is to go to the SNOM's webpage, and under Setup/Advanced set Call join on Xfer (2 calls) to off Why that option is enabled by default I don't know, but it caused me grief for a couple of months until I finally discovered that option - and even now the receptionist refuses to answer more than one call at a time, because she's afraid to join two customers together again. - Matt ___ 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] cvs problems?
Hi, I am trying to call a number with bv, and when ever I try to bridge a call it gets this message. Meanwhile it works fine if I use a softphone. Is this broken? Or I am making some mistake? Regards, Dinesh. owl:/usr/src# asterisk -r Asterisk CVS-HEAD-03/07/05-17:14:42, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk CVS-HEAD-03/07/05-17:14:42 currently running on owl (pid = 184) Verbosity is at least 3 -- Executing Dial(SIP/10.217.84.12-40704778, SIP/broadvoice/19784187300) in new stack -- Called broadvoice/19784187300 -- SIP/broadvoice-45bf is ringing -- SIP/broadvoice-45bf answered SIP/10.217.84.12-40704778 -- Attempting native bridge of SIP/10.217.84.12-40704778 and SIP/broadvoice-45bf -- Got SIP response 404 Not Found back from 147.135.8.128 -- Got SIP response 404 Not Found back from 147.135.8.128 -- Got SIP response 404 Not Found back from 147.135.8.128 -- Got SIP response 404 Not Found back from 147.135.8.128 -- Got SIP response 404 Not Found back from 147.135.8.128 -- Got SIP response 404 Not Found back from 147.135.8.128 -- Got SIP response 404 Not Found back from 147.135.8.128 -- Got SIP response 404 Not Found back from 147.135.8.128 -- Got SIP response 404 Not Found back from 147.135.8.128 -- Got SIP response 404 Not Found back from 147.135.8.128 == Spawn extension (default, 219784187300, 1) exited non-zero on 'SIP/10.217.84.12-40704778' owl*CLI exit ___ 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] ISDN to SIP
Hello.
If I receive a Phone call by ISDN or from SIP Provider, the Asterisk make
some errors and the SIP Client don't react.
The messages from Asterisk in verbose mode:
er will net.
Asterisk messages in Terminalmode:
parse_srv: SRV mapped to host sip-ha.web.de, port 5060
Mar 10 00:02:17 NOTICE[5776]: chan_sip.c:7191 handle_request: Failed to
authenticate user unknown sip:[EMAIL PROTECTED];tag=as5bfdabe6
Mar 10 00:02:17 NOTICE[5776]: chan_sip.c:7191 handle_request: Failed to
authenticate user unknown sip:[EMAIL PROTECTED];tag=as76a8acb1
Mar 10 00:02:17 NOTICE[5776]: chan_sip.c:7191 handle_request: Failed to
authenticate user unknown sip:[EMAIL PROTECTED];tag=as29a2f623
Mar 10 00:02:18 NOTICE[5776]: chan_sip.c:7191 handle_request: Failed to
authenticate user unknown sip:[EMAIL PROTECTED];tag=as44aed266
-- parse_srv: SRV mapped to host sip-ha.web.de, port 5060
-- creating pipe for PLCI=0x101 msn = 456456
-- started pbx on channel (callgroup=0)!
== Starting CAPI[contr1/456456]/3 at ,5480080,1 failed so falling back to
exten 's'
== Starting CAPI[contr1/456456]/3 at ,s,1 still failed so falling back to
context 'default'
Mar 10 00:04:42 WARNING[5776]: pbx.c:1882 ast_pbx_run: Channel
'CAPI[contr1/456456]/3' sent into invalid extension 's' in context
'default', but no invalid handler
-- Executing Hangup(CAPI[contr1/456456]/3, ) in new stack
== Spawn extension (default, h, 1) exited non-zero on 'CAPI[contr1/456456/3'
-- CAPI Hangingup
-- removed pipe for PLCI = 0x101
Here is my sip.conf:
[general]
bindaddr = 0.0.0.0
port = 5060
context = default
maxexpirey = 3600
defaultexpirey = 120
srvlookup = yes
tos = 0x18
disallow = all
allow = gsm
allow = alaw
allow = ulaw
allow = g729
register = christoph:[EMAIL PROTECTED]/christoph.hehl
[web_de]
context = default
type = friend
host = sip.web.de
username = christoph
secret = password
fromuser = christoph
fromdomain = sip.web.de
dtmfmode = inband
nat = yes
insecure = no
[chris]
type = friend
secret = passwd
host = dynamic
dtmfmode = rfc2833
nat = no
callerid = chris 11
canreinvite = no
qualify = no
insecure = very
my extensions.conf
static = yes
writeprotect = no
[globals]
[default]
exten = h,1,Hangup
exten = 11,1,Dial(SIP/chris,,tr)
exten = 11,2,Hangup
exten = 456456,1,Dial(11,,tr)
exten = 456456,2,Hangup
exten = _0.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],,tr)
exten = _0.,2,Hangup
exten = _1.,1,Dial(CAPI/@456456:${EXTEN:1},,tr)
exten = _1.,2,Hangup
Please Help
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Re: [Asterisk-Users] Apple links Asterisk
On Mar 10, 2005, at 6:31, Matthew Boehm wrote: From macintouch.com: Apple is distributing an open-source Asterisk install package for Mac OS X: I suppose they get a little overexcited. Apple isn't distributing anything, they just link to a third party that made a ready-to-install package. That link has been up since August 2004, and the Asterisk version it uses is CVS 10-28-03... yikes :) I might be interesting to build from a recent source and extract the extra pieces they advertise out of that installer package. jens ___ 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 NOTIFY problem
Hi, I am not getting the NOTIFY messages from asterisk though other UA un-REGISTERS with asterisk. What is the reason? Is there any thing that we *MUST* configure for this? Please do help me in this regard. With regards Somesh S. Shanbhag --- somesh s [EMAIL PROTECTED] wrote: Hi, I have this scenario. UA1 == AS UA2 UA1 : User Agent 1 UA2 : User Agent 2 AS : Asterisk AS has been configured with UA1 UA2 users. Registrations are happening correctly. But.. UA1 AS == UA2 SUBSCRIBE to UA2 --- 200 OK NOTIFY 200 OK--- As shown above the subscriptions are also proper I am getting the open status correctly for UA2. But when UA2 unregisters then asterisk is NOT issuing any NOTIFY to UA1 with closed status. Why this is happening? Is there any thing I missed in configuring? Please help me in this regard With warm regards Somesh S. Shanbhag --- SIMPLICITY IS THE BEAUTY. BE NATURAL LIVE NATURAL. --- Somesh S. Shanbhag Mascon Global Communication Technologies Enterprise of Mascon Global Limited #59/2, 100Ft Ring Road Banashankari II stage Bangalore-560070 Karnataka INDIA Website: http://www.masconit.com --- __ Do you Yahoo!? Yahoo! Sports - Sign up for Fantasy Baseball. http://baseball.fantasysports.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 --- SIMPLICITY IS THE BEAUTY. BE NATURAL LIVE NATURAL. --- Somesh S. Shanbhag Mascon Global Communication Technologies Enterprise of Mascon Global Limited #59/2, 100Ft Ring Road Banashankari II stage Bangalore-560070 Karnataka INDIA Website: http://www.masconit.com --- __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ 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: Zap channels intermittently bridging withSNOM190
Hi Matt, Thanks for your reply, greatly appreciated ! I will log into the phone shortly and check it out. I will be very relieved if it fixes my problem. because she's afraid to join two customers together again. Yea, my client is starting to get that way too ! :) I'll let you know if I come right - thank you for your assistance so far. Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion ! ___ Computers are not intelligent. They only think they are. - Original Message - From: Matt Kemner [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 10, 2005 10:21 AM Subject: Re: [Asterisk-Users] OT: Zap channels intermittently bridging withSNOM190 Hi David On Thu, 10 Mar 2005, quoth David Wilson: Something I've picked up with the SNOM is that sometimes when there are two active incoming calls via Zap channels they end up being transferred to eachother. In other words two in bound callers end up being connected to eachother. I've come across this exact problem. Whenever there are two calls on hold and you push the transfer button, it transfers them both together. The solution is to go to the SNOM's webpage, and under Setup/Advanced set Call join on Xfer (2 calls) to off Why that option is enabled by default I don't know, but it caused me grief for a couple of months until I finally discovered that option - and even now the receptionist refuses to answer more than one call at a time, because she's afraid to join two customers together again. - Matt ___ 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] Apple links Asterisk
Matthew Boehm wrote: From macintouch.com: Apple is distributing an open-source Asterisk install package for Mac OS X: A complete IP-PBX in software. SNIP If anyone's interested, Benjamin Kowarsch from Sunrise Telephone systems Ltd is doing that. Check it out at http://www.sunrise-tel.com You can also google the mailing list for his email, if interested. Flynn ___ 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: OT: Best DB
Forrest W. Christian [EMAIL PROTECTED] writes: Today, MySQL has more features that it lacked earlier - i.e. it's got transactions and additional queries, and so on. I understand that PostgreSql has also gotten faster than it used to be. So, at this point it's almost devolved into a holy war as opposed to there being any real difference. Emphasis on almost, though. MySQL still has a long way to go, and the bits that are missing or inferior will take an awful lot of work to catch up. It *is* getting better with every release, though. It's just still (as it has been throughout) trailing quite a bit behind PostgreSQL as a real RDBMS. Its huge popularity was a matter of timing and luck, and once it reached critical mass, well... :-) That said, lots and lots of people are quite happy with their MySQL installations. The reason why I recommend not choosing MySQL for a new project is that you don't know when you'll suddenly need some capability that it doesn't have -- and finding out after you've invested a lot of time and effort isn't any fun. Better to use what is known to have everything you need *plus* most of the stuff you might conceivably find yourself needing in the future. -tih -- Don't ascribe to stupidity what can be adequately explained by ignorance. ___ 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: Zap channels intermittently bridging withSNOM190
Hi Matt, I've looked but can't find that option in the Settings-Advanced. I've got other options such as: CMC Feature: on off Dialog-Info Call Pickup: on off Call Waiting Indication: on off Dialtone during Hold: on off Disconnect on Hook: on off etc. But not Call join on Xfer (2 calls). Perhaps a difference in firmware ? Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion ! ___ Computers are not intelligent. They only think they are. - Original Message - From: Matt Kemner [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 10, 2005 10:21 AM Subject: Re: [Asterisk-Users] OT: Zap channels intermittently bridging withSNOM190 Hi David On Thu, 10 Mar 2005, quoth David Wilson: Something I've picked up with the SNOM is that sometimes when there are two active incoming calls via Zap channels they end up being transferred to eachother. In other words two in bound callers end up being connected to eachother. I've come across this exact problem. Whenever there are two calls on hold and you push the transfer button, it transfers them both together. The solution is to go to the SNOM's webpage, and under Setup/Advanced set Call join on Xfer (2 calls) to off Why that option is enabled by default I don't know, but it caused me grief for a couple of months until I finally discovered that option - and even now the receptionist refuses to answer more than one call at a time, because she's afraid to join two customers together again. - Matt ___ 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] Cisco 7940/60 and 802.3af PoE
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Are any versions of the Cisco 7940/7960 or 7940G/7960G phones compatible with the 802.4af Power over Ethernet Standard? Ok, I know the question has been asked before, but googling has turned up several contradictory results: 1/ No - not at all 2/ Maybe - 79XXG will work 3/ With a special cable/dongle (a la wikki) I am looking at getting several 20 x 7960 (not Gs) to work with * and a NetGear FSM7326P switch. Do I also need to get a PowerDSine converter dongle for each phone? TIA - -- Ron Wellsted http://www.wellsted.org.uk [EMAIL PROTECTED] FWD: 519961 N 52.567623, W 2.137621 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.6 (GNU/Linux) iQEVAwUBQjAYlEtP/KMNOfRbAQK/PQgAseiWaL9WD7EEbsXVitARMHch4ewVyJva WyOW7HqV1wh/kQj8RPAANyIwDuLcBUiJ1SzMLZeNM7qq4YEAuvqua8mFUlh/VjnR yA0RkIM83im54RZQzYELwUOGtWH0znbdlGJc6qFoGgNAkA9BBA/pBmYrZb2syGgX IrbMhQTSlPs8hE8i/GbFJkubfCfEO+3g7Pgp11SuLrDz1enSFX/KcsXTd89kgvEP QYckM2kiRFnx0APNWsHrukj2giapO/Gu7XhvW8fCkBDmaajFuxZzdBSHB6VrMObX 1dJG0zzfcE0Q98+NRMqpqZrfMSlTJ2f+xhRIQpdbaeS9U6QEKQzn5A== =Gk6/ -END 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
RE: [Asterisk-Users] Print-to-Fax client
This conversation is interesting, What about a driver that will send the print out to Asterisk, on the same network to be sent as Fax ? Is there anything that already exists for this? Quoting Florian Overkamp [EMAIL PROTECTED]: Hi, -Original Message- You should be able to download one (for WIndows and possibly Mac) from efax or j2.com I think. http://www.efax.com/en/efax/twa/page/download?rqcp=2 http://www.j2.com/jconnect/twa/page/download You might be able to do that, but take a good look at the license agreement on the driver - you might not be allowed to use the software fully without having a subscription to their services. Florian ___ 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 This message was sent using IMP, the Internet Messaging Program. ___ 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] Which hardware for this solution?
Frank Sautter wrote: if you want to splice asterisk between a pbx and an S0 (ISDN-BRI) from your telco then you will need a ISDN Card that support NT mode e.g. cards with HFC chipset like those from www.junghanns.net or www.beronet.com. Thank you very much, Frank! Googling around with those new terms :-) as NT mode, HFC chipset ISDN-BRI I found this site http://isdn.jolly.de/ That's the homepage of pbx4linux project, but it has a nice list of ISDN cards capable of NT mode. It may be useful to others, who knows... That's a pity that Digium does not provied any ISDN card. (and please correct me if I spotted a wrong list, for my purposes) Thanks again, Giorgio Mandolfo ___ 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] ODBC error ?
Am Donnerstag 10 März 2005 00:55 schrieb Jens Kübler: If someone could explain me this (added some extra debugging code Mar 10 00:47:13 NOTICE[7561]: chan_sip.c:8448 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '172.20.41.7' Mar 10 00:47:13 WARNING[7561]: res_config_odbc.c:97 realtime_odbc: SQL param name is a7 Mar 10 00:47:13 WARNING[7561]: res_odbc.c:90 odbc_smart_execute: SQL Execute error! Attempting a reconnect... Mar 10 00:47:13 WARNING[7561]: res_odbc.c:411 odbc_obj_disconnect: res_odbc: disconnected 0 from pgsql [PostgreSQL-asterisk] Mar 10 00:47:13 NOTICE[7561]: res_odbc.c:468 odbc_obj_connect: Connecting pgsql Mar 10 00:47:13 NOTICE[7561]: res_odbc.c:483 odbc_obj_connect: res_odbc: Connected to pgsql [PostgreSQL-asterisk] Mar 10 00:47:13 WARNING[7561]: res_config_odbc.c:104 realtime_odbc: SQL Execute error! [SELECT * FROM sip_hadiko WHERE name = ?] Problem was: GRANT rights ON table TO user Jens ___ 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] Server specifications
Callum, Will you be having any VoIP phones connected to the system, or will it just be the E1s? If so, how many do you expect, how many calls do you expect, and what codecs will you be using? Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Callum McGillivray wrote: Hi all, Can someone point me to some information on the type of hardware that might / should be used for a high load on an asterisk machine ? I know that this is dependant on what services you plan to have running, and its relevant to what you plan to do. We are likely to be running 4 E1s, Voicemail, IVR menus, Music on Hold, Pay-Over-The-Phone, lots of interdepartmental calls and will probably have 60 channels from the E1s running concurrently, with the possibility of all 120 being used during high load periods. Im trying to gauge what kind of hardware we might be looking at, and whether the system should be split across multiple servers, but Im having a hard time finding anything solid. Cheers, Callum ___ 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] Cisco 7960
Hi There I am currently having an issue with a Cisco 7960. The phone is using SIP firmware version 6.3. I have successfully got the phone to register with Asterisk and I can call the phone from other non Cisco handsets. However when I dial out from the 7960 I do not even see any output on the Asterisk console. Is there some sort of DTMF setting which I might have incorrectly set ? Following DTMF settings are in my SIPdefault.cnf file. dtmf_inband: 1 dtmf_outband: avt dtmf_db_level: 3 It seems to me like Asterisk is not detecting any key tones from the phone. I have followed a number of setup guides for this phone to no avail. Any help or suggestions are greatly appreciated. Regards Ed ___ 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] Grandstream Message button
Hi I confiured Gasnstream phone 100. Firmware ver:Program--1.0.5.16 Bootloader--1.0.0.21 HTML--1.0.0.41 ïVOC--1.0.0.7. It workes well everything. If I got a message it blinks. My voicemail no 555 .If I call 555,I can hear voicemail . But I can not configure Message Button on the phone. I set via html Voice Mail UserID:555. If I press message button does not work. Can you help me ? Thanks. Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more.___ 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: Zap channels intermittently bridging withSNOM190
Hi David I've looked but can't find that option in the Settings-Advanced. But not Call join on Xfer (2 calls). Perhaps a difference in firmware ? Could be, I'm running snom190-SIP 3.56m - if you're not running at least that firmware already you should upgrade. I upgraded it to fix another problem we were having, which is that call-waiting would not work if you had the phone off-hook but were not actually talking to someone (eg just put someone on hold, or in the process of dialing) - the phone would report busy to asterisk, and would never make the call-waiting beep. I emailed Snom who were very helpful - they replied within hours, and told me a fix would be out soon - which it was within a few days. btw the subject of this post is a little misleading - I had the call join problem with ISDN and SIP calls as well, not just Zap. - Matt ___ 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] Calls hang in a conversation
Hello guys, I have an Asterisk installed (last version), and sometimes I have my calls hanged when I'm still talking to someone. I put the maximum debug to have informations about this problem and I found only one thing : Mar 10 01:26:16 DEBUG[16136]: Requesting Hangup because the busy tone was detected on channel Zap/1-1 Mar 10 01:26:16 DEBUG[16136]: Got a FRAME_CONTROL (5) frame on channel Zap/1-1 Mar 10 01:26:16 DEBUG[16136]: Bridge stops bridging channels SIP/remi-615d and Zap/1-1 Mar 10 01:26:16 DEBUG[16136]: Set option AUDIO MODE, value: ON(1) on Zap/1-1 Mar 10 01:26:16 DEBUG[16136]: Hangup: channel: 1 index = 0, normal = 18, callwait = -1, thirdcall = -1 Mar 10 01:26:16 DEBUG[16136]: Not yet hungup... Calling hangup once with icause, and clearing call Do you have any idea ? Thanks. Jean-Philippe Le Henaff [EMAIL PROTECTED] ___ 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: Zap channels intermittently bridging withSNOM190
Hi Matt, Thanks for your reply. Ah excellent ! Ok. I'm running snom190-SIP 3.44. I'll upgrade to the latest version and see what happens. I'm sure it will sort out my problems. Sorry about the misleading subject :) I started a couple days ago being very unclear about how things were going wrong and thought it could be something in Asterisk that was causing it. Thanks so much for all your help. Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion ! ___ Computers are not intelligent. They only think they are. - Original Message - From: Matt Kemner [EMAIL PROTECTED] To: David Wilson [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 10, 2005 12:40 PM Subject: Re: [Asterisk-Users] OT: Zap channels intermittently bridging withSNOM190 Hi David I've looked but can't find that option in the Settings-Advanced. But not Call join on Xfer (2 calls). Perhaps a difference in firmware ? Could be, I'm running snom190-SIP 3.56m - if you're not running at least that firmware already you should upgrade. I upgraded it to fix another problem we were having, which is that call-waiting would not work if you had the phone off-hook but were not actually talking to someone (eg just put someone on hold, or in the process of dialing) - the phone would report busy to asterisk, and would never make the call-waiting beep. I emailed Snom who were very helpful - they replied within hours, and told me a fix would be out soon - which it was within a few days. btw the subject of this post is a little misleading - I had the call join problem with ISDN and SIP calls as well, not just Zap. - Matt ___ 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: Active channels bridging with SNOM190
Yea, True. No sweat. Should be better now ? :-) Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion ! ___ Computers are not intelligent. They only think they are. - Original Message - From: Matt Kemner [EMAIL PROTECTED] To: David Wilson [EMAIL PROTECTED] Sent: Thursday, March 10, 2005 12:57 PM Subject: Re: [Asterisk-Users] OT: Zap channels intermittently bridging withSNOM190 On Thu, 10 Mar 2005, quoth David Wilson: Sorry about the misleading subject :) I started a couple days ago being very unclear about how things were going wrong and thought it could be something in Asterisk that was causing it. Yeah I know what you mean.. I specifically didn't contact SNOM about this bug because I also had this nagging feeling that it could be an asterisk config problem, and I didn't want to hassle them about it if it was. I only made the comment about the subject in case someone in the future comes across this problem and looks in the archives, just so they're not put off thinking it's a different bug. - Matt ___ 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] Tired of trying to fix this echo problem
Martin Roy wrote: snip I'm tired of beeing unable to get rid correctly of the echo problem. I have 3 TDM04B installed in one server. We had to adjust the [rx | tx}gain settings in zapata.conf for a couple of phones to get rid of the echo. Most is gone. you might try setting the tx to a less than zero db value while keeping rx at zero for starters. g ___ 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] Cisco 7960
Modify the dialplan.xml on your tftp server to this DIALTEMPLATE TEMPLATE MATCH=* Timeout=4 User=Phone/ /DIALTEMPLATE Jason On Thu, 10 Mar 2005 10:31:34 -, Marshall, Ed [EMAIL PROTECTED] wrote: Hi There I am currently having an issue with a Cisco 7960. The phone is using SIP firmware version 6.3. I have successfully got the phone to register with Asterisk and I can call the phone from other non Cisco handsets. However when I dial out from the 7960 I do not even see any output on the Asterisk console. Is there some sort of DTMF setting which I might have incorrectly set ? Following DTMF settings are in my SIPdefault.cnf file. dtmf_inband: 1 dtmf_outband: avt dtmf_db_level: 3 It seems to me like Asterisk is not detecting any key tones from the phone. I have followed a number of setup guides for this phone to no avail. Any help or suggestions are greatly appreciated. Regards Ed ___ 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] Grandstream Message button
asterisk asterisk wrote: Hi I confiured Gasnstream phone 100. Firmware ver:Program--1.0.5.16 Bootloader--1.0.0.21HTML--1.0.0.41 ï VOC--1.0.0.7. The message button under 1.0.5.16 was broken, go to 1.0.5.18 or newer to fix. Doug ___ 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] FWDout credits sharing
This post is a FWDout specific one, but may be of interest for many Asterisk users and may even make more of them use FWDout for the good of everyone :) On the one hand, I have seen many reports of people using the FWDout (http://www.fwdout.net/) service who don't get credits because the prefix they provide calls for to is not popular. On the other hand, I get plenty of unused credits, much more than I can use. Although I didn't try it because I don't need more credits, cheating to obtain credits is technically very easy. And by the way, if you need credits, just send me your phone number complete with country (via private email, not on, this list), make sure you provide a specific route to it in FWDout so that your system gets used, and I may call you in order to increase your credits balance. I think the credits system should be totally reorganized, this is what appears to me as one of the weakest part of FWDout. For example, it could be something like: - receive 5 credits every day when your system is up - allow credits transfer from one account to another Also, being able to rate the quality of the various routes that have been used by a user would be very helpful in selecting the best one. Sam -- Samuel Tardieu -- [EMAIL PROTECTED] -- http://www.rfc1149.net/sam ___ 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] Grandstream Message button
There's apparently a bug in the latest firmware that knackers the Message button, but Grandstream haven't fixed it yet. A right pain indeed. Stuart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk asterisk Sent: 10 March 2005 10:33 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Grandstream Message button Hi I confiured Gasnstream phone 100. Firmware ver:Program--1.0.5.16 Bootloader--1.0.0.21HTML--1.0.0.41 o VOC--1.0.0.7. It workes well everything. If I got a message it blinks. My voicemail no 555 .If I call 555, I can hear voicemail . But I can not configure Message Button on the phone. I set via html Voice Mail UserID:555. If I press message button does not work. Can you help me ? 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
[Asterisk-Users] Problem with NOTIFY
Hi, Consider the scenario Asterisk Version : 1.0.5 UA1 = Asterisk Server (AS) = UA2 UA1 UA2 have successfully registered with AS (Asterisk Server). Now UA1 sends SUBSCRIBE (for UA2) to AS UA1 gets 200 OK to SUBSCRIBE. AS sends NOTIFY with open status to UA1 for UA2. Now if UA2 unregisters (REGISTER with expires = 0), then no NOTIFY is issued to UA1 about UA2. UA1 assumes UA2 as still in open status. But AS *MUST* have sent NOTIFY with status closed to UA1. Why this is not happening? Is there any configuration problem? Are there any special commands in asterisk to enable notifications like this?? Please do help in this regard With regards Firdosh ___ 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: how to sip-h323 using asterisk-oh323-0.7.1
HELLO
i am using gungk gatekeeper from a provider. he has
given me a account,password,ip now i want to connect
to it with asterisk.
1. i want to call to my sip phones registered on my
local area network working. ok
2. i want to divert PSTN call to gun gatekeeper (from
service provider company). not working
the problem is that when i am trying to connect it
asterisk is desplaying message that Gatekeeper
'gatekeeper ip' found but faild to register.
i am using asterisk-oh323-0.7.1. one thing more when i
am using there diler to connection its working fine.
oh323.conf
---
[general]
listenAddress=myip
listenPort=1720
connectPort=1720
tcpStart=1
tcpEnd=2
udpStart=1
udpEnd=2
fastStart=yes
h245Tunnelling=no
h254inSetup=no
inBandDTMF=yes
silenceSupperession=no
jitterMin=20
jitterMax=100
ipTos=none
tos=lowdelay
outboundMax=10
inboundMax=10
simultaneousMax=10
wrapLibTraceLevel=1
libTraceLevel=1
libTraceFile=stdout
gatekeeper=provider ip
accountcode=account from provider
gatekeeperPassword=account password from provider
gatekeeperTTL=600
userUnputMode=TONE
amaFlags=default
context=default
[register]
context=default
alias=666
[665]
type=h323
prefix=321
context=default
codec=G711U
frames=20
extensions.conf
--
[default]
exten=2000,1,Dial(SIP/${EXTEN})
exten=3000,1,Dial(SIP/${EXTEN})
exten=_321X,1,Dial(OH323:h323/[EMAIL PROTECTED]:1720|30|r)
2000, 3000 is working with
i want 32145671 now my call should be transfered to
provider and dial his number 45671
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RE: [Asterisk-Users] how do i get rid of this blasted echo !!!
Sorry for the delay in replying to this post, my hardware platform is an HP Compaq D530 SFF - 2.4gb /512mb /40gb. I have a TDM02b configuration (2xFXO). You mention FCC and the mode for UK impedance, can you explain a little further on this one as I am a little lost, I cant see any setting with FCC anywhere (even in the config samples) so could you explain whereabouts I set this ? The other question I have is regarding the MMX stuff you talk about. My processor is a P4 2.4ghz, excuse my lack of knowledge here but I thought MMX was a feature bundled with processors of about 5 years ago?? Do I still need to recompile with MMX support, and if so, some pointer on how to do this would be appreciated (also, can I recompile the zaptel drivers and carry on as normal, or will I need to reinstall asterisk again?) Any help would be greatly appreciated Regards Gary -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: 03 March 2005 12:32 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] how do i get rid of this blasted echo !!! On March 3, 2005 07:02 am, Brett, Gary wrote: I have 2 TDM400P's, 2 asterisk servers (running on powerful boxes with FC1 and * v CVS 1.0.02), and 4 analogue PSTN lines from BT and whatever I do, I cannot get rid of this damn local echo. Ive tried setting the echoTraining, echoCancel (in phone.conf and Zapata.conf) , echocancelwhenbridged to every possible combination , Ive even tried running the fxotune utility to no avail. Ive swapped cards, telephone lines, servers and also tried different phones (budgetone, x-lite, 7940) but still it's the same. You haven't told us what hardware (platform) you're on, nor have you told us if your FXO ports are in whatever mode they need to be in for UK impedances (I think they default to FCC or North American). For echo on my PRI I could not get rid of it until I recompiled the zaptel and wct4xxp drivers with MMX support enabled and with the instructions reordered and used for the pentium 4 processor (which I'm using, Xeon 2.6 to be exact). After that, the echo magically disappeared. I haven't reverted back to my original (non-processor-optimized, non-MMX-enabled) drivers to see if it comes back, but that's all that's changed and it's in production so I am hesitant to screw around with it any more. -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 ___ 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] Grandstream Message button
On Thu, 2005-03-10 at 11:55 +, Stuart Ford wrote: There's apparently a bug in the latest firmware that knackers the Message button, but Grandstream haven't fixed it yet. A right pain indeed. FWIW 1.0.5.22, which I believe to be the latest, works A1 on my 15 phones. -- Dave Cotton [EMAIL PROTECTED] ___ 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 and other ISA T1 cards
On March 9, 2005 11:36 pm, David Josephson wrote: There is a mention that the current Sangoma T1 cards (A10[1,2,4]) work with * using their WANPIPE drivers. Has anyone used any older Sangoma cards that also support WANPIPE ? I imagine Sangoma would have this answer for you. :-) -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
Re: [Asterisk-Users] Grandstream Message button
On Thu, 10 Mar 2005 13:32:13 +0100, Dave Cotton [EMAIL PROTECTED] wrote: On Thu, 2005-03-10 at 11:55 +, Stuart Ford wrote: There's apparently a bug in the latest firmware that knackers the Message button, but Grandstream haven't fixed it yet. A right pain indeed. FWIW 1.0.5.22, which I believe to be the latest, works A1 on my 15 phones. Indeed - the bug was fixed in 1.0.5.18 (Nov 14 2004) according to the changelog. Fixed BT-100 dialing bad URI when using the message button Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ 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] Compiling Asterisk On SUSE 9.2
Dear all, I have tried to compile * 1.0.6 (downloaded from the digium site, in the right sequence - zaptel, libpri, asterisk) on two different machines running SUSE 9.2. The problem comes during some preliminary checks: checking for ar... /usr/bin/ar checking for tgetent in -ltermcap... no checking for tgetent in -ltinfo... no checking for tgetent in -lcurses... no checking for tgetent in -lncurses... no configure: error: termcap support not found make: *** [editline/libedit.a] Error 1 Now I got the termcap rpm and afaik it's installed (now). Is there anything obvious I should try? Thanks in advance, Aldo ___ 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: Active channels bridging with SNOM190
Hi Matt, Cool. The upgrade went through well and the Call join on Xfer option appeared. I've now turned the option off and so far things are working nicely. Thank you so much for your help. It looks like this issue has been sorted ! Keep well. Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion ! ___ Computers are not intelligent. They only think they are. - Original Message - From: David Wilson [EMAIL PROTECTED] To: Matt Kemner [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 10, 2005 1:08 PM Subject: Re: [Asterisk-Users] OT: Active channels bridging with SNOM190 Yea, True. No sweat. Should be better now ? :-) Kindest regards David Wilson ___ D c D a t a Tel +27 33 342 7003 Fax +27 33 345 4155 Cell +27 82 4147413 http://www.dcdata.co.za [EMAIL PROTECTED] Powered by Linux, driven by passion ! ___ Computers are not intelligent. They only think they are. - Original Message - From: Matt Kemner [EMAIL PROTECTED] To: David Wilson [EMAIL PROTECTED] Sent: Thursday, March 10, 2005 12:57 PM Subject: Re: [Asterisk-Users] OT: Zap channels intermittently bridging withSNOM190 On Thu, 10 Mar 2005, quoth David Wilson: Sorry about the misleading subject :) I started a couple days ago being very unclear about how things were going wrong and thought it could be something in Asterisk that was causing it. Yeah I know what you mean.. I specifically didn't contact SNOM about this bug because I also had this nagging feeling that it could be an asterisk config problem, and I didn't want to hassle them about it if it was. I only made the comment about the subject in case someone in the future comes across this problem and looks in the archives, just so they're not put off thinking it's a different bug. - Matt ___ 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] FWDout credits sharing
This post is a FWDout specific one, but may be of interest for many Asterisk users and may even make more of them use FWDout for the good of everyone :) Sam, some people have come to the conclusion that while FWDOut is a ince idea, it isn't a good idea. The first two things that come to mind are the Letting other use you phone line is very likely against the contract you have with your provider or telco and could get you sued or cancelled. This is a very real issue and will loom closer as voIP puts a bigger dent in their revenues. Having calls you don't make come through your line, the number of which can be identified by law enforcement in the case of a crime or harassment, or gee, suspected terrorist activities which some countries are very sensitive about... Those were enough for me to reconsider and remove Bellster from my server. ___ 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] Print-to-Fax client
What about a driver that will send the print out to Asterisk, on the same network to be sent as Fax ? Is there anything that already exists for this? Hello, Several months ago I worked on such a solution using salsafax. The problem was on how to determine the fax number to send the fax to. I tried with OCR but had a 60% success rate extracting the number. It was cool for me but not good for a bussiness. FYI, salsafax is a script for use with Samba and CUPS/Lpr. Basically you export a printer to the network, and then you can setup that network printer in your windows/samba clients and print to it. Then you have to convert the postrcript file to .tiff to be used by txfax. Another problem is that I do not know if spandsp can return the status of the fax after it is sent, so you know if it was received ok or not. Regards, Quoting Florian Overkamp [EMAIL PROTECTED]: Hi, -Original Message- You should be able to download one (for WIndows and possibly Mac) from efax or j2.com I think. http://www.efax.com/en/efax/twa/page/download?rqcp=2 http://www.j2.com/jconnect/twa/page/download You might be able to do that, but take a good look at the license agreement on the driver - you might not be allowed to use the software fully without having a subscription to their services. Florian -- Nicolás Gudiño Buenos Aires - Argentina ___ 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] Echo for first 15 to 20 seconds
Not sure what the default value is, but seems to me it was something like 100 (for = yes). If I recall correctly from the days when this was implemented, it represents a delay in milliseconds from the time a zap channel is seized until the echotraining sequence is initiated. For my system with a TDM04b, =800 does a good job. Try different values on your system to see what the minimum value is that improves your echo issues. I thought echotraining=400 was the default? Cheers, Dean -Original Message- I am using asterisk with a handful of DM04B cards. Everything seems fine except for an echo on all calls on the local end of the call. In almost all cases the echo goes away after 15 to 20 seconds. I am attributing the echo going away to the echo cancellation code that was enabled when the following options are set: echocancel=yes ; echocancelwhenbridged=yes echotraining=yes ; Instead of echotraining=yes, use echotraining=800 and don't forget to 'stop' and restart asterisk. A simple reload won't cut it. ___ 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: voicepulse silence during conversations
Sean Kennedy [EMAIL PROTECTED] wrote: Hi all, I'm running Asterisk 1.0.0. I am a customer ( and supporter ) of voicepulse. For me, it works perfectly, but one of my customers noticed a small problem: During a conversation, when the otherside isn't talking, it's almost like the mic turns off. I have noticed this too, especially when speaking to someone who is using a cell phone. I assume that VP is using silence suppression. -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.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] New Integrics tip: VoIP for ISPs
All, I've posted a new tip on the Integrics website. It's on how ISPs can offer VoIP service to their customers, and why it makes good business sense to do so. http://integrics.com/tips/voip_for_isps/ Older tips can be found at: http://integrics.com/tips/ -- Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.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] Compiling Asterisk On SUSE 9.2
Aldo Bergamini wrote: Dear all, I have tried to compile * 1.0.6 (downloaded from the digium site, in the right sequence - zaptel, libpri, asterisk) on two different machines running SUSE 9.2. The problem comes during some preliminary checks: checking for ar... /usr/bin/ar checking for tgetent in -ltermcap... no checking for tgetent in -ltinfo... no checking for tgetent in -lcurses... no checking for tgetent in -lncurses... no configure: error: termcap support not found make: *** [editline/libedit.a] Error 1 Now I got the termcap rpm and afaik it's installed (now). Is there anything obvious I should try? Thanks in advance, Aldo ___ 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 you need to install ncurses-dev ___ 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] Problem with incoming calls.
Title: Problem with incoming calls. I have a problem with incoming IAX calls. I have 2 numbers from the same supplier delivered over IAX. I register once with the server, and both calls get to my box, and I get output on the console with both calls. However I cannot get each number to go to separate contexts Please see relevant sections from extensions and iax conf. files Section from IAX.conf: [448700XX] ;incoming 0870 number type=user username=448700XX context=conference trunking=off [448450XX] ;incoming 0845 number type=user username=448450XX context=demo_default trunking=off Section from extensions.conf [demo_default] ;the 0845 number should go here exten = 448450XX,1,Answer exten = ..i have more here. [conference] ;the 0870 number should go here exten = 448700XX,1,Answer exten = ..i have more here. The output on the CLI looks like: NOTICE[1282]: chan_iax2.c:5461 socket_read: Rejected connect attempt from XXX.XXX.X.XXX, request '[EMAIL PROTECTED]' does not exist However if you look above, the 0870 number should go to the [conference] context; not the [demo_default] one.. If I then call the other 0845 number it works: -- Accepting unauthenticated call from XXX.XXX.X.XXX, requested format = 8, actual format = 8 -working If I comment out one of the numbers in the iax.conf, the other one works fine.its just when both are active it doesnt seem to play properly. Does anyone have any ideas? As far as I know I'm not being stupid, but please point it out if I am. Any help much appreciated. Regards, C ___ 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] Compiling Asterisk On SUSE 9.2
On Thursday 10 March 2005 12:47, Aldo Bergamini wrote: Dear all, I have tried to compile * 1.0.6 (downloaded from the digium site, in the right sequence - zaptel, libpri, asterisk) on two different machines running SUSE 9.2. The problem comes during some preliminary checks: checking for ar... /usr/bin/ar checking for tgetent in -ltermcap... no checking for tgetent in -ltinfo... no checking for tgetent in -lcurses... no checking for tgetent in -lncurses... no configure: error: termcap support not found make: *** [editline/libedit.a] Error 1 Now I got the termcap rpm and afaik it's installed (now). Is there anything obvious I should try? It's telling you that you have no curses devel package installed. B ___ 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: Best DB
On Thu, Mar 10, 2005 at 01:04:35AM -0700, Forrest W. Christian said: I understand that PostgreSql has also gotten faster than it used to be. It's interesting. Just yesterday I was saying that we use both MySQL and Postgres here, and that we were probably going to move everything to postgres just to consolidate. Now one of our lead engineers has done some performance testing last night for our app and found MySQL to be 8 to 100 times faster for all but one of our operations (combination of ~80% reads, 20% writes on the InnoDB table type.) His testing basically increased the load until performance was unacceptable. This is with lots of optimizations on Postgres (the current DB for the app) and none on MySQL. Needless to say, we now need to re-evaluate our decision to move everything to Postgres. In the end, it all comes down to knowing exactly what features you need for your app, how your specific app performs on each DB, what you need for support, etc. As Forrest mentioned, write DB independant code and then you can easily choose the DB that is best for your app. 2 years for now, you may find a need to switch DB's. ___ 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] BRI: Unable to create channel of type 'ZAP'
Hi, I'm trying to build a SIP / ISDN BRI gateway. I'm using asterisk and zaptel 1.0.6 with the bristuff patches. I have a Billion HFC card connected to a BRI ISDN line. Unfortunately each time asterisk tries to make a call on that channel, I get NOTICE[6236]: app_dial.c:759 dial_exec: Unable to create channel of type 'ZAP' == Everyone is busy/congested at this time When I try to call from the outside, all I get is: chan_zap.c:7786 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 The module zaphfc is loaded correctly. Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 266-399 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. On startup, Asterisk seems to initialize ZAP correctly: [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, PRI Signalling signalling -- Registered channel 2, PRI Signalling signalling -- Automatically generated pseudo channel == Starting D-Channel on span 1 == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI) == Registered channel type 'Tor' (Zapata Telephony Driver w/PRI) I'm not sure why it says PRI though. Here's my configuration: zaptel.conf === span=1,1,2,ccs,ami bchan=1-2 dchan=3 defaultzone=se loadzone=se zapata.conf === signalling=bri_cpe echocancel=yes echocancelwhenbridged=yes group=1 callerid=asreceived context=frompstn prilocaldialplan=unknown pridialplan = unknown channel = 1-2 Thanks, Johan. ___ 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: Grandstream Message button
Peter Bowyer [EMAIL PROTECTED] wrote: Indeed - the bug was fixed in 1.0.5.18 (Nov 14 2004) according to the changelog. Is this a beta version of the firmware? The main download page only has 1.0.5.16. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.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] Comparison Charts
I couldnt agree with you more Jim. Im realdy using Asterisk and agree 100% with what you say... I was asking for a comparison list with other PBX's because for example, for a customer, they have heard of Avaya and Cisco and they all are selling IP now... So In order to get your customer to trust Asterisk over those guys, you need to show him the diff. Between the two and some lists of the features on the others compared to Asterisk.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Van Meggelen Sent: Jueves, 10 de Marzo de 2005 12:17 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Comparison Charts [EMAIL PROTECTED] wrote: Guys. Anybody has a URL or some document with comparison charts with Asterisk's features against other PBXs? I would argue that what you ask is in some ways impossible. Asterisk is orders of magnitude more flexible than any other PBX you may have encountered, because it is more like a toolkit than a PBX. Whatever is missing can be built, so there's no list of features that can ever be considered complete. For people who are looking for a PBX that has a user-friendly interface and is easy to configure, Asterisk will tend to dissappoint. Where Asterisk shines is for those people who want to--need to--build their own PBX. People who are willing to do the work themselves; designing, testing, debugging, re-designing . . . Many of us believe that Asterisk is going to transform the telecommunication industry, but it won't do it because it has more features, it'll do it because it puts the control of the features list where it belongs: in the customer's hands. I would suggest that the best way to approach Asterisk is to have a list of things that you need your telephone system to do. Then, one-by-one, figure out how to handle each of those in Asterisk. When you are done, you may have a few that you couldn't find a satisfactory solution to. Those can typically be custom developed, and surprisingly, you will still probably come in at a lower cost than a closed, so-called full-featured proprietary system. What's more, as your needs grow, Asterisk can grow with you. Five years from now you won't need to hear oh sorry but that system is no longer supported. Want new functionality? Install it. Is the hard drive wearing out? Replace it. Need more CPU power? Migrate to a new chassis. Asterisk changes all the rules. Therfore, to understand it, you have to adopt a new way of thinking about telecom systems. Welcome to Asterisk! -- Jim Van Meggelen [EMAIL PROTECTED] -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.0 - Release Date: 08/03/2005 ___ 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 ___ 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] Where can I find all areacodes for USA (accountingpurpose)
On Thu, 10 Mar 2005 15:25:01 +0800, Ronald Wiplinger [EMAIL PROTECTED] wrote: Jim Van Meggelen wrote: I prefer to have an expression like: 1[1-7] .18 are toll free, 19 are premium rate numbers, ... That's not going to work... all of South Carolina (1-803, 1-843, 1-864), parts of Virginia (1-804), Utah (1-801), California (1-818), Pennsylvania (1-814), etc, would be toll-free. Parts of Florida (1-904), North Carolina (1-910, 1-919), New York (1-917), etc would be premium rate calls. You're going to need the full list from NANPA if you want your call logs to make any sense. tg. ___ 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] a liitle bit of info required
Hi I am fairly new to Asterisk. I will have few questions on one Asterisk system: Description: I looked at Asterisk Mall says that The TrueLine SMB PBX is perfect for the small office providing service for up to 28 Telco lines. This is a 1U Rack Mount Single T1 Asterisk Appliance. Question: Does it mean it has 7 PCI bus and takes 7x4=28 channels (if they used 7 TDM400P card with 4 FXOs on each), then only they can use this Asterisk box with SIP or H323 phones, no analog/digital phones. Is this right? Question: I believe TDM400P is full length (with FXO or FXS cards, without, it is half length) card. How does it fit to 1u rack with FXOs on it? Or Am I confused with 1U rack size? cheers Turgut ___ 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] Problems with new install voicemail broadcast
Tried that as well, same result.. Rick From: Eric_Doiron [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 09, 2005 11:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Problems with new install voicemail broadcast Try specifying the contexts ... just an idea exten = 1,4,VoiceMail([EMAIL PROTECTED][EMAIL PROTECTED]) -E From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rick Harby Sent: Wednesday, March 09, 2005 5:53 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Problems with new install voicemail broadcast Have a fair amount of asterisk experience, but this one is blowing my mind.. I have a context setup as follows: [department-listing] exten = s,1,Background(custom/6000) exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,30 ; exten = 1,1,Answer exten = 1,2,Wait(1) exten = 1,3,Background(pls-wait-connect-call) exten = 1,4,VoiceMail(u620122) exten = 1,5,Hangup So basically, when they press 1, it should leave voicemail in box 620 and 122 Here is my voicemail conf: [default] 122=2301,Sales Guy,,,tz=eastern 620=2301,General Sales Mailbox,,,delete=yes|tz=eastern Ive modified the VoiceMail account to just me Voicemail(u620) and it works fine, and Ive tried Voicemail(u122) and it works, but when this is in there, and someone does 1 this is the what happens: Executing BackGround(Zap/2-1, pls-wait-connect-call) in new stack -- Playing 'pls-wait-connect-call' (language 'en') -- Executing VoiceMail(Zap/2-1, u620122) in new stack Mar 9 17:33:08 WARNING[-1396851792]: app_voicemail.c:1518 leave_voicemail: No entry in voicemail config file for '620122' Executing Hangup(Zap/2-1, ) in new stack Any ideas would be greatly appreciated. ___ 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: Best DB
Walt Reed wrote: On Thu, Mar 10, 2005 at 01:04:35AM -0700, Forrest W. Christian said: I understand that PostgreSql has also gotten faster than it used to be. It's interesting. Just yesterday I was saying that we use both MySQL and Postgres here, and that we were probably going to move everything to postgres just to consolidate. Now one of our lead engineers has done some performance testing last night for our app and found MySQL to be 8 to 100 times faster for all but one of our operations (combination of ~80% reads, 20% writes on the InnoDB table type.) His testing basically increased the load until performance was unacceptable. This is with lots of optimizations on Postgres (the current DB for the app) and none on MySQL. Needless to say, we now need to re-evaluate our decision to move everything to Postgres. In the end, it all comes down to knowing exactly what features you need for your app, how your specific app performs on each DB, what you need for support, etc. As Forrest mentioned, write DB independant code and then you can easily choose the DB that is best for your app. 2 years for now, you may find a need to switch DB's. Out of curiosity, what version of PostgreSQL was used? 7.x, 8.x? Also, was the test run on the same system? I'm not looking to bash. I'm just curious as we are in the same MySQL/PostgreSQL boat. David ___ 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: Grandstream Message button
Indeed - the bug was fixed in 1.0.5.18 (Nov 14 2004) according to the changelog. Is this a beta version of the firmware? The main download page only has 1.0.5.16. And the phones are downloading 1.0.5.16 via TFTP from 168.75.215.189 - is there somewhere else they should be looking? Stuart ___ 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] Location of Voice e-mail Code???
Hi, Where can I find the code that performs the voice e-mail function (that is, the code that reads the contents of voicemail.conf and then performs the necessary action)? I am using [EMAIL PROTECTED] 0.6. Thanks in advance! -- Rgds, Julius. ___ 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] Print-to-Fax client
On Thu, 10 Mar 2005, Nicolás Gudiño wrote: What about a driver that will send the print out to Asterisk, on the same network to be sent as Fax ? Is there anything that already exists for this? For HylaFax several adapter programs exist for Windows. See e.g. http://support.real-time.com/open-source/hylafax/win2k/index.html. At least the Python-based programs should be adaptable to collect the information and send to asterisk. Alternativly, you can wait until SpanDSP is able to talk to HylaFax directly. Peter ___ 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] how do i get rid of this blasted echo !!!
On Thu, 10 Mar 2005 12:29:32 -, Brett, Gary [EMAIL PROTECTED] wrote: Sorry for the delay in replying to this post, my hardware platform is an HP Compaq D530 SFF - 2.4gb /512mb /40gb. I have a TDM02b configuration (2xFXO). You mention FCC and the mode for UK impedance, can you explain a little further on this one as I am a little lost, I cant see any setting with FCC anywhere (even in the config samples) so could you explain whereabouts I set this ? The other question I have is regarding the MMX stuff you talk about. My processor is a P4 2.4ghz, excuse my lack of knowledge here but I thought MMX was a feature bundled with processors of about 5 years ago?? Do I still need to recompile with MMX support, and if so, some pointer on how to do this would be appreciated (also, can I recompile the zaptel drivers and carry on as normal, or will I need to reinstall asterisk again?) MMX is a feature that is bundled since the early days of PentiumII (around 1997), it has never been removed, and is still with every pentium cpu from intel. To enable MMX in zaptel, before you compile zaptel, uncomment the line that says: /* #define CONFIG_ZAPTEL_MMX */ and change it to: #define CONFIG_ZAPTEL_MMX Any help would be greatly appreciated Regards Gary -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: 03 March 2005 12:32 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] how do i get rid of this blasted echo !!! On March 3, 2005 07:02 am, Brett, Gary wrote: I have 2 TDM400P's, 2 asterisk servers (running on powerful boxes with FC1 and * v CVS 1.0.02), and 4 analogue PSTN lines from BT and whatever I do, I cannot get rid of this damn local echo. Ive tried setting the echoTraining, echoCancel (in phone.conf and Zapata.conf) , echocancelwhenbridged to every possible combination , Ive even tried running the fxotune utility to no avail. Ive swapped cards, telephone lines, servers and also tried different phones (budgetone, x-lite, 7940) but still it's the same. You haven't told us what hardware (platform) you're on, nor have you told us if your FXO ports are in whatever mode they need to be in for UK impedances (I think they default to FCC or North American). For echo on my PRI I could not get rid of it until I recompiled the zaptel and wct4xxp drivers with MMX support enabled and with the instructions reordered and used for the pentium 4 processor (which I'm using, Xeon 2.6 to be exact). After that, the echo magically disappeared. I haven't reverted back to my original (non-processor-optimized, non-MMX-enabled) drivers to see if it comes back, but that's all that's changed and it's in production so I am hesitant to screw around with it any more. -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 ___ 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: Grandstream Message button
On Thu, 2005-03-10 at 09:48 -0400, Doug Meredith wrote: Is this a beta version of the firmware? The main download page only has 1.0.5.16. What's beta software? We're told 90% of the world is using it and paying for it. http://www.grandstream.com/BETATEST/ -- Dave Cotton [EMAIL PROTECTED] ___ 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] Cisco 7960
On Thu, 10 Mar 2005 10:31:34 -, Marshall, Ed [EMAIL PROTECTED] wrote: Hi There I am currently having an issue with a Cisco 7960. The phone is using SIP firmware version 6.3. I have successfully got the phone to register with Asterisk and I can call the phone from other non Cisco handsets. However when I dial out from the 7960 I do not even see any output on the Asterisk console. Is there some sort of DTMF setting which I might have incorrectly set ? Following DTMF settings are in my SIPdefault.cnf file. dtmf_inband: 1 dtmf_outband: avt dtmf_db_level: 3 A sip phone using the SIP protocol never sends actual DTMF when dialing, instead it uses an invite message to asterisk, which has the digits dialed in it. The tones you hear -when pushing the buttons on your phone- are actualy not sent to asterisk until you either press dial or #, or if your dialplan setup in diaplan.xml sends it. The inband and outband settings are only used when already talking on the phone (I might be wrong on this one, anybody out there if I am, please let me know). What does the phones display tell you when you dial? It seems to me like Asterisk is not detecting any key tones from the phone. I have followed a number of setup guides for this phone to no avail. Any help or suggestions are greatly appreciated. Regards Ed ___ 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] Cisco 7940/60 and 802.3af PoE
It worked for me without any special cables (I'm using 7960, and not 7960G). I'm using a netgear POE switch (FSM7326PNA NETGEAR). If for you it doesn't work as is, the cable mods on the wiki should help you. On Thu, 10 Mar 2005 09:51:11 + (GMT), Ron Wellsted [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Are any versions of the Cisco 7940/7960 or 7940G/7960G phones compatible with the 802.4af Power over Ethernet Standard? Ok, I know the question has been asked before, but googling has turned up several contradictory results: 1/ No - not at all 2/ Maybe - 79XXG will work 3/ With a special cable/dongle (a la wikki) I am looking at getting several 20 x 7960 (not Gs) to work with * and a NetGear FSM7326P switch. Do I also need to get a PowerDSine converter dongle for each phone? TIA - -- Ron Wellsted http://www.wellsted.org.uk [EMAIL PROTECTED] FWD: 519961 N 52.567623, W 2.137621 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.6 (GNU/Linux) iQEVAwUBQjAYlEtP/KMNOfRbAQK/PQgAseiWaL9WD7EEbsXVitARMHch4ewVyJva WyOW7HqV1wh/kQj8RPAANyIwDuLcBUiJ1SzMLZeNM7qq4YEAuvqua8mFUlh/VjnR yA0RkIM83im54RZQzYELwUOGtWH0znbdlGJc6qFoGgNAkA9BBA/pBmYrZb2syGgX IrbMhQTSlPs8hE8i/GbFJkubfCfEO+3g7Pgp11SuLrDz1enSFX/KcsXTd89kgvEP QYckM2kiRFnx0APNWsHrukj2giapO/Gu7XhvW8fCkBDmaajFuxZzdBSHB6VrMObX 1dJG0zzfcE0Q98+NRMqpqZrfMSlTJ2f+xhRIQpdbaeS9U6QEKQzn5A== =Gk6/ -END 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 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: Best DB
On March 10, 2005 08:44 am, Walt Reed wrote: Now one of our lead engineers has done some performance testing last night for our app and found MySQL to be 8 to 100 times faster for all but one of our operations (combination of ~80% reads, 20% writes on the InnoDB table type.) His testing basically increased the load until performance was unacceptable. I'd *love* to see the particulars of that test. It's been shown time and time again that postgres' speed CLOBBERS mysql for anything but the simplest selects, and that it can handle far more concurrent connections without slowing down. Have you asked the folks on freenode #postgresql as well? This is with lots of optimizations on Postgres (the current DB for the app) and none on MySQL. Needless to say, we now need to re-evaluate our decision to move everything to Postgres. It's quite possible your optimizations are buggering things up too. I ran into that. :-) -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
Re: [Asterisk-Users] a liitle bit of info required
On Thu, 10 Mar 2005 15:53:05 +0200 Turgut Abacioglu [EMAIL PROTECTED] wrote: Hi I am fairly new to Asterisk. I will have few questions on one Asterisk system: Description: I looked at Asterisk Mall says that The TrueLine SMB PBX is perfect for the small office providing service for up to 28 Telco lines. This is a 1U Rack Mount Single T1 Asterisk Appliance. Question: Does it mean it has 7 PCI bus and takes 7x4=28 channels (if they used 7 TDM400P card with 4 FXOs on each), then only they can use this Asterisk box with SIP or H323 phones, no analog/digital phones. Is this right? Question: I believe TDM400P is full length (with FXO or FXS cards, without, it is half length) card. How does it fit to 1u rack with FXOs on it? Or Am I confused with 1U rack size? cheers Turgut Why don't you try http://www.asteriskmall.com/aboutus.asp and use their phone number or email address to find out directly from the company??? You will probably get a much quicker and more accurate response since they are the ones building the system. ___ 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] Delay on outgoing calls
Hi, I've a wildcard TDM400P card with 2 fxo and 2 fxs modules. I've set this extension in my extensions.conf for obtain the external line: exten = 0,1,Dial(Zap/g2,10) The dial application is executed immediatly but next this there is a delay before I can hear the tone. This is the output in the CLI: -- Starting simple switch on 'Zap/1-1' -- Executing Dial(Zap/1-1, Zap/g2|10) in new stack -- Called g2 -- Zap/3-1 answered Zap/1-1 -- Attempting native bridge of Zap/1-1 and Zap/3-1 The delay is about 2 seconds and it is between Called g2 and Zap/3-1 answered Zap/1-1. Where can I reduce this delay? I have tried to find it in zaptel.h, but I'm not sure if it is in the driver sources or in the asterisk sources. Thank you in advance, Stefano Arata Italy ___ 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: Grandstream Message button
Doug Meredith wrote: Peter Bowyer [EMAIL PROTECTED] wrote: Indeed - the bug was fixed in 1.0.5.18 (Nov 14 2004) according to the changelog. Is this a beta version of the firmware? The main download page only has 1.0.5.16. Yes, the current is 1.0.5.22, but I've found it to be very stable. http://www.grandstream.com/BETATEST ___ 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: Compiling Asterisk On SUSE 9.2
[EMAIL PROTECTED] is believed to have said: you need to install ncurses-dev Thanks! Aldo ___ 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: Grandstream Message button
- Original Message - From: Stuart Ford [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 10, 2005 8:02 AM Subject: RE: [Asterisk-Users] Re: Grandstream Message button Indeed - the bug was fixed in 1.0.5.18 (Nov 14 2004) according to the changelog. Is this a beta version of the firmware? The main download page only has 1.0.5.16. And the phones are downloading 1.0.5.16 via TFTP from 168.75.215.189 - is there somewhere else they should be looking? Stuart Try here: http://gs-firmware.gratissip.dk/firmwares/ ___ 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: Compiling Asterisk On SUSE 9.2
[EMAIL PROTECTED] is believed to have said: It's telling you that you have no curses devel package installed. B Thanks! Aldo ___ 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 busy message every couple of days.
I have my broadvoice asterisk server up and running. For some reason after every couple of days you call the number and it says the number you are trying to reach is busy and cannot take your call right now. I then stop asterisk and start it and it is fine for a couple days. Has anyone else had this issue? Any idea why? Randy ___ 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: Best DB
IB/FB stores the DB in one file, but the file can span multiple drives if needed. However, you can't select which table goes into which file. Personally, I don't think that's very feasible, nor is it required -- if a table is accessed often enough to be mission critical, large parts of it will reside in memory due to caching anyway. -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Sent: Thursday, March 10, 2005 1:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] OT: Best DB If it stores the entire DB in 1 file, it can not scale as well as other DBs. Postgres 8 supports splitting a single DB up so you can put portions of it on different media if needed. If you have to tune for absolute speed, you can purchase one of the solid state drives for the tables that need that kind of speed while using much less expensive harddrives for the rest of the DB. While I do not remember mysql supporting it this directly, I think I remember the file structure being not to difficult to figure out and split and symlink back together if need be. ___ 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: Grandstream Message button
Try here: http://gs-firmware.gratissip.dk/firmwares/ Trouble is, I've got dozens of GS handsets on many sites all downloading firmware from 168.75.215.189 - as advised on the Grandstream website :) Stuart ___ 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] ASTCC - regexpression for country and certain cities?
How do I key in the Regex pattern for certain cities and other country parts. E.g., China 86 China Beijing 8610, 8611, 8612, 8614 ~ 8619 ^86 ^86[0-2,4-9] I am not sure how to set up both expressions. Can anybody give me a hand for that? bye Ronald ___ 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] Asterisk-oh323-0.7.1 compile error
Hi; I use the following asterisk, openh323, pwlib: asterisk = cvs-head-03/09/05 openh323 = 1.13.5 pwlib = 1.6.6 asterisk-oh323= 0.7.1 Asterisk, openh323, pwlib were compiled successfully but when I try to compile Asterisk-Oh323-0.7.1, I got the following error: chan_oh323.o chan_oh323.cchan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory. ... . . ... make[1]: *** [chan_oh323.o] Error 1make[1]: Leaving directory `/root/asterisk-oh323-0.7.1/asterisk-driver'make: *** [subdirs_build] Error 1 It seems , it looks for "channel_pvt.h" which I coudnot find it in Asterisk source.Should I use different CVS-Head?? Apperciate your helps Mohammad ___ 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] MP3 stream for MOH
CJ Toma wrote: Any suggestions how can I get asterisk to play MOH (music on hold) a MP3 radio stream from the internet (http:// location) instead of a MP3 file in the mphmp3 folder? I tried putting default = quietmp3:http://www.waixwave.com/pacnet.pls instead of default = quietmp3:/var/lib/asterisk/mohmp3 but did not work got message NOTICE[25564]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! Any suggestions how to get the mp3 stream work? Thanks. CJ http://www.voip-info.org/wiki-Asterisk Has several examples. ie.. http://www.voip-info.org/tiki-index.php?page=Using%20Slimserver%20for%20Music%20on%20Hold slimp3 = custom:/var/lib/asterisk/mohmp3-dummy,/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://localhost:9000/stream.mp3 But once you get it going, it doesn't work anyway. Asterisk stops MOH (closes the stream) when channel hangs up. This is great for all other MOH uses, but drops the mp3 stream and doesn't reconnect to streaming sever. (as noted in original patch/bug #413) Unless someones worked on fixing this. I was streaming XM radio thru MOH. But for now, your better off just moving along. ___ 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] NuFone
Anyone know how many simultaneous calls you can receive on a NuFone DID? -Mark ___ 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: Best DB
On Thu, Mar 10, 2005 at 08:55:53AM -0500, David Filion said: Walt Reed wrote: Now one of our lead engineers has done some performance testing last night for our app and found MySQL to be 8 to 100 times faster for all but one of our operations (combination of ~80% reads, 20% writes on the InnoDB table type.) His testing basically increased the load until performance was unacceptable. This is with lots of optimizations on Postgres (the current DB for the app) and none on MySQL. Needless to say, we now need to re-evaluate our decision to move everything to Postgres. Out of curiosity, what version of PostgreSQL was used? 7.x, 8.x? Also, was the test run on the same system? I'm not looking to bash. I'm just curious as we are in the same MySQL/PostgreSQL boat. We are useing 7.4.6. Considering 8.0 just came out in January, and considering how many major changes went into it, we were leary of upgrading until it had time to get tested by the masses. I would expect 8.x to be faster that 7.x, but I didn't see anything in the release notes that would indicate a 1 to 2 orders of magnitude performance increase. The tests were run on the same server (RHEL3 on a maxed out DL380-g4). We had been tuning the table design / query design, postgres config, etc. for quite some time, trying to get better performance. the mysql install was just the standard binaries available on the mysql site, with the default config. ___ 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] Sipura-841 Problems
Hello Everyone. Please let me know if this is the incorrect forum for this question. I recently acquired 3 Sipura-841 phones for use with my asterisk system. I provisioned and deployed one. It worked for 3 days, then uddenly, it stopped working. Upon reboot, the MWI light blinks 20 times and that's the end of it. It has never registered again since then. So I replaced the phone with another that I had. I provisioned it, it went online for a split second, then it too died! MWI light blinks 20 times, and the phone is dead. So Sipura sent me out 3 replacement phones the next week. Now, I've provisioned a replacement phone. I was talking on it for 2 hours when it died. (guess what) Same problem mostly. MWI light blinks 20 times, and the phone just doesn't work. (Except this one will reboot itself every 5 minutes. About every 10th reboot, it will go online for a few minutes, then die again) Curious if anyone has experienced the same problem? I have tried all these phones in four totally unique environments. (ie: differetn dhcp servers, differnet hubs/switches, different ISPs, etc) Problem is consistent across them all. I've tried cvs-head, and asterisk-1.05, same problem. One other thing.. If I use the exact same SIP credentials/config on a Linksys PAP2, or Linksys RT31P2, it works fine, or even any softphone. Any direction would be appreciated. Les ___ 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] NuFone
On Mar 10, 2005, at 7:06 AM, Mark Halverson wrote: Anyone know how many simultaneous calls you can receive on a NuFone DID? If you're talking about an 800-number DID, then I don't think there's a limit, at least until you start saturating their capacity. There have been reports of users with 20+ calls going at once during an emergency. I don't know if the same rules apply to their Michigan DIDs; I think they're using a fixed-price model for them, which usually means that you're limited to 1 or 2 calls at a time. Scott ___ 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: Best DB
I'd *love* to see the particulars of that test. It's been shown time and time again that postgres' speed CLOBBERS mysql for anything but the simplest selects, and that it can handle far more concurrent connections without slowing down. I strongly agree with this, i have a prepaid voip solution with asterisk, freeradius and postgresql , the hole thing relies in stored procedures and triggers (i mean the billing, traffic monitoring, admin system, etc). It had scaled from thousands of minutes per month to two millions in these days without an issue, we export the cdr to mysql for the IT/Customer Service guys, because they have php/mysql programmers (We do Java/Postgres), but we keep the original data in our postgres DB, for simple select like the sum of minutes per month or per customer in a period of time, yeah, postgres is about 30% slower than mysql, but if they want to know the total minutes/calls per destination country, customers and peak hours, we run a single stored procedure with temporal tables and cursors, wait some seconds and... oh yeah, the beautiful report suddently appears in our screen, then we smile while our workmates run 3 or 4 querys (stored procedures in mysql?, not yet, triggers?, not yet, cursors?, not yet, what else ?? not yet,etc) and wait minutes for their results. Hardware HP DL380, 2x 2.8 Ghz, 3GB RAM Software Asterisk, Freeradius, Postgresql 7.4, Mysql 4.1 Just my $0.02 --- Miguel ___ 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] hide callerid via presention bits of ISDN
Hi, how can I setup asterisk to use the number presentation bits on the isdn side to suppress the number presentation? We need to transmit the subscriber number for billing purposes via ISDN whether or not the user wants to hide his/her number. Is there any way to do this? Deti ___ 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] Cisco 7940/60 and 802.3af PoE
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Are any versions of the Cisco 7940/7960 or 7940G/7960G phones compatible with the 802.4af Power over Ethernet Standard? Ok, I know the question has been asked before, but googling has turned up several contradictory results: 1/ No - not at all 2/ Maybe - 79XXG will work 3/ With a special cable/dongle (a la wikki) I'm using a 3COM POE injector with the cable modifications found on the wiki and it works great. Calvin -- S i x F e e t U p | Nowhere to go but open-source Silicon Valley: +1 (650) 401-8579 | Midwest: +1 (317) 861-5948 Toll-Free: 1-866-SIX-FEET mailto:[EMAIL PROTECTED] http://www.sixfeetup.com | Zope Hosting from $19.95/month ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Avoiding connect signal in two stage dialing
Hello,
I am using asterisk for two stage dialing:
1-) I make a call from my voip phone;
2-) Asterisk dials first a access number, iputs a PIN and then dials destination
number;
However, I am getting the connect signal from the moment access number connects.
I would like to avoid receiving this signal and only get connect signal after
the destination number connects.
Any clue how to implement this?
I have been using this:
exten -- _. , 1, Dial(SIP/myprovider.com, 30,
D(www447881234ww12345678www${EXTEN})
This message was sent using IMP, the Internet Messaging Program.
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RE: [Asterisk-Users] Sipura-841 Problems
Wow... sort of a horror story there. Not sure I have a lot of input other then we have two 841's that we've been using for about a month now without out problems. (I've actually been rather impressed.) One thing that comes to mind though, have you tried updating the firmware to the latest? Or is that impossible giving their behavior.. -James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of LES.NET (1996) INC. Sent: Thursday, March 10, 2005 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Sipura-841 Problems Hello Everyone. Please let me know if this is the incorrect forum for this question. I recently acquired 3 Sipura-841 phones for use with my asterisk system. I provisioned and deployed one. It worked for 3 days, then uddenly, it stopped working. Upon reboot, the MWI light blinks 20 times and that's the end of it. It has never registered again since then. So I replaced the phone with another that I had. I provisioned it, it went online for a split second, then it too died! MWI light blinks 20 times, and the phone is dead. So Sipura sent me out 3 replacement phones the next week. Now, I've provisioned a replacement phone. I was talking on it for 2 hours when it died. (guess what) Same problem mostly. MWI light blinks 20 times, and the phone just doesn't work. (Except this one will reboot itself every 5 minutes. About every 10th reboot, it will go online for a few minutes, then die again) Curious if anyone has experienced the same problem? I have tried all these phones in four totally unique environments. (ie: differetn dhcp servers, differnet hubs/switches, different ISPs, etc) Problem is consistent across them all. I've tried cvs-head, and asterisk-1.05, same problem. One other thing.. If I use the exact same SIP credentials/config on a Linksys PAP2, or Linksys RT31P2, it works fine, or even any softphone. Any direction would be appreciated. Les ___ 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] Re: OT: Best DB
Walt Reed [EMAIL PROTECTED] writes: I would expect 8.x to be faster that 7.x, but I didn't see anything in the release notes that would indicate a 1 to 2 orders of magnitude performance increase. A few points concerning PostgreSQL and performance: - Each of the latest releases has improved performance quite a bit. - Out of the box, it is tuned for minimal resource use, and dismal performance. It really needs to be tuned. Check out Josh Berkus's web site, http://www.powerpostgresql.com/, for hints and tips. - Nothing helps much if your schema and your queries are suboptimal. Think about how your data is used, consider what indexes you need, rewrite slow queries to be smarter (use EXPLAIN). - Did I mention you need to tune the database system to your needs? -tih -- Don't ascribe to stupidity what can be adequately explained by ignorance. ___ 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] how do i get rid of this blasted echo !!!
Hi all, I'm still fighting echo on a T1-PRI too. I see the suggestion about enabling MMX in zaptel/zconfig.h but I didn't understand the bit about reordering the instructions. Can you elaborate? Thanks, Jon - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 10, 2005 9:07 AM Subject: Re: [Asterisk-Users] how do i get rid of this blasted echo !!! On Thu, 10 Mar 2005 12:29:32 -, Brett, Gary [EMAIL PROTECTED] wrote: Sorry for the delay in replying to this post, my hardware platform is an HP Compaq D530 SFF - 2.4gb /512mb /40gb. I have a TDM02b configuration (2xFXO). You mention FCC and the mode for UK impedance, can you explain a little further on this one as I am a little lost, I cant see any setting with FCC anywhere (even in the config samples) so could you explain whereabouts I set this ? The other question I have is regarding the MMX stuff you talk about. My processor is a P4 2.4ghz, excuse my lack of knowledge here but I thought MMX was a feature bundled with processors of about 5 years ago?? Do I still need to recompile with MMX support, and if so, some pointer on how to do this would be appreciated (also, can I recompile the zaptel drivers and carry on as normal, or will I need to reinstall asterisk again?) MMX is a feature that is bundled since the early days of PentiumII (around 1997), it has never been removed, and is still with every pentium cpu from intel. To enable MMX in zaptel, before you compile zaptel, uncomment the line that says: /* #define CONFIG_ZAPTEL_MMX */ and change it to: #define CONFIG_ZAPTEL_MMX Any help would be greatly appreciated Regards Gary -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: 03 March 2005 12:32 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] how do i get rid of this blasted echo !!! On March 3, 2005 07:02 am, Brett, Gary wrote: I have 2 TDM400P's, 2 asterisk servers (running on powerful boxes with FC1 and * v CVS 1.0.02), and 4 analogue PSTN lines from BT and whatever I do, I cannot get rid of this damn local echo. Ive tried setting the echoTraining, echoCancel (in phone.conf and Zapata.conf) , echocancelwhenbridged to every possible combination , Ive even tried running the fxotune utility to no avail. Ive swapped cards, telephone lines, servers and also tried different phones (budgetone, x-lite, 7940) but still it's the same. You haven't told us what hardware (platform) you're on, nor have you told us if your FXO ports are in whatever mode they need to be in for UK impedances (I think they default to FCC or North American). For echo on my PRI I could not get rid of it until I recompiled the zaptel and wct4xxp drivers with MMX support enabled and with the instructions reordered and used for the pentium 4 processor (which I'm using, Xeon 2.6 to be exact). After that, the echo magically disappeared. I haven't reverted back to my original (non-processor-optimized, non-MMX-enabled) drivers to see if it comes back, but that's all that's changed and it's in production so I am hesitant to screw around with it any more. -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 ___ 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
Re: [Asterisk-Users] Asterisk-oh323-0.7.1 compile error
try with a CVS head before 03/03/05 as the channel structure was changed then, or get an updated version of asterisk-oh323 if there is one availiable Jason On Thu, 10 Mar 2005 06:25:04 +0330, mohammad [EMAIL PROTECTED] wrote: Hi; I use the following asterisk, openh323, pwlib: asterisk = cvs-head-03/09/05 openh323 = 1.13.5 pwlib = 1.6.6 asterisk-oh323= 0.7.1 ___ 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] hide callerid via presention bits of ISDN
On Thu, 10 Mar 2005 16:22:39 +0100, Deti Fliegl [EMAIL PROTECTED] wrote: Hi, how can I setup asterisk to use the number presentation bits on the isdn side to suppress the number presentation? We need to transmit the subscriber number for billing purposes via ISDN whether or not the user wants to hide his/her number. Is there any way to do this? look at this extension.conf command show application SetCallerPres asterisk*CLI -= Info about application 'SetCallerPres' =- [Synopsis]: Set CallerID Presentation [Description]: SetCallerPres(presentation): Set Caller*ID presentation on a call. Always returns 0. Valid presentations are: allowed_not_screened: Presentation Allowed, Not Screened allowed_passed_screen : Presentation Allowed, Passed Screen allowed_failed_screen : Presentation Allowed, Failed Screen allowed : Presentation Allowed, Network Number prohib_not_screened : Presentation Prohibited, Not Screened prohib_passed_screen: Presentation Prohibited, Passed Screen prohib_failed_screen: Presentation Prohibited, Failed Screen prohib : Presentation Prohibited, Network Number unavailable : Number Unavailable ___ 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] bypassing auth info
Hello everyone, I have the following problem with forwarding the authentication info, hope you will know a solution. I need to connect few remote asterisks to one in my location via IAX, and then forward all the calls initiated by them via SIP or H323 to PSTN cisco gateway or SER proxy server. Unfortunatelly, all the authentication and accounting need to be done on this terminating gateway or SER server. So, what I need to do is to bypass calls on my asterisk to SER or cisco with usernamepassword data of the IAX originator of a call, not of my asterisk server. All the calls would go to one destination (SER or Cisco) but from the different originators, and they need to be distinguished on terminating destination. Also, the best would be to allow everyone to connect via IAX to my asterisk, without a need to update iax.conf/extensions.conf every time. AAA would be done on terminating gateway/proxy. Any idea how this could be done ? thanks, Argon _ On the road to retirement? Check out MSN Life Events for advice on how to get there! http://lifeevents.msn.com/category.aspx?cid=Retirement ___ 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: OT: Best DB
On Thu, Mar 10, 2005 at 04:39:25PM +0100, Tom Ivar Helbekkmo said: Walt Reed [EMAIL PROTECTED] writes: I would expect 8.x to be faster that 7.x, but I didn't see anything in the release notes that would indicate a 1 to 2 orders of magnitude performance increase. A few points concerning PostgreSQL and performance: - Each of the latest releases has improved performance quite a bit. - Out of the box, it is tuned for minimal resource use, and dismal performance. It really needs to be tuned. Check out Josh Berkus's web site, http://www.powerpostgresql.com/, for hints and tips. - Nothing helps much if your schema and your queries are suboptimal. Think about how your data is used, consider what indexes you need, rewrite slow queries to be smarter (use EXPLAIN). - Did I mention you need to tune the database system to your needs? You snipped out my paragraph where I mentioned tuning the DB itself, queries, and schema. I have no doubt that 8.x is faster than 7.x, but I did not find any reports from people claiming a 10X performance boost. I didn't look hard, but I did look. I'll look into installing 8.x and see if we can rerun the tests. ___ 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] tdm400p and dell 2600 poweredge
Hi all: I've been developing and testing on a tdm400p card and it's been going well. As you probably know, the tdm400p needs an ide power supply, but the dell poweredge 2600 that this card is destined for eventually has all the power supplied on the backplane with no ide cables. The thing is, on the motherboard in the server, there is an ide ribbon connector, and beside that, something also marked ide, that looks suspiciously like a power supply takeoff. I've talked to the good folks at dell about this, and they are as clueless as ever. Has anyone run into this problem before and, if so, have you found the secret cable that turns the power supply on the board into a regular ide cable? Cheers, -grant -- Grant McInnes [EMAIL PROTECTED] ___ 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] NVFaxDetect errors on make
Hi All, I am trying to add FAX to my SIP confiig and I am getting some errors, any help would be great. gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DASTERISK_VERSION=\"CVS-v1-0-12/23/04-22:36:11\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN -fPIC -c -o app_nv_faxdetect.o app_nv_faxdetect.capp_nv_faxdetect.c: In function `nv_detectfax_exec':app_nv_faxdetect.c:210: error: structure has no member named `cid'app_nv_faxdetect.c:227: error: structure has no member named `cid'app_nv_faxdetect.c:265: error: structure has no member named `cid'make[1]: *** [app_nv_faxdetect.o] Error 1make[1]: Leaving directory `/usr/src/asterisk/apps'make: *** [subdirs] Error 1linux01:/usr/src/asterisk # Thanks, Chris TuskaNetwork EngineerCCNA, CCSA In theory, theory and practice are the same. In practice, they aren't ___ 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: how to sip-h323 using asterisk-oh323-0.7.1
hello
now i am using my own gnugatekeeper. asterisk is
registering successfully with Gnugatekeeper. but it is
not transfering call to gnugk.
i am running 1234 user of OpenPhone with GNUgatekeeper
when i try to call from sip User agent 3000 to 3211234
asterisk is not forwarding it to GnuGK it replying
with 404 not found.
gatekeeper.ini
[Gatekeeper::Main]
Fourtytwo=42
TimeToLive=600
[RoutedMode]
GKRouted=1
H245Routed=0
CallSignalPort=1721
[RasSrv::PermanentEndpoints]
asterisk mechine ip=xyz;123
[GkStatus::Auth]
rule=allow
on asterisk
oh323.conf
---
[general]
listenAddress=myip
listenPort=1719
connectPort=1719
tcpStart=1
tcpEnd=2
udpStart=1
udpEnd=2
fastStart=yes
h245Tunnelling=no
h254inSetup=no
inBandDTMF=yes
silenceSupperession=no
jitterMin=20
jitterMax=100
ipTos=none
tos=lowdelay
outboundMax=10
inboundMax=10
simultaneousMax=10
wrapLibTraceLevel=1
libTraceLevel=1
libTraceFile=stdout
gatekeeper=gnugk ip
accountcode=account
;gatekeeperPassword=account password
gatekeeperTTL=600
userUnputMode=TONE
amaFlags=default
context=default
[XYZ]
type=h323
prefix=123
context=default
codec=G711U
frames=20
extensions.conf
--
[default]
exten=2000,1,Dial(SIP/${EXTEN})
exten=3000,1,Dial(SIP/${EXTEN})
exten=_123,1,Dial(SIP/${EXTEN})
exten=_321,1,Dial(OH323:h323/[EMAIL PROTECTED]:1719|30|r)
sip.conf
--
[2000]
host=dynamic
type=friend
dtmfmode=INFO
canreinvite=no
[3000]
host=dynamic
type=friend
dtmfmode=INFO
canreinvite=no
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RE: [Asterisk-Users] Problem with incoming calls.
Title: Problem with incoming calls. I now have a workaround for this problem; I carried on research after posting. I now route both incoming numbers in the IAX.conf into one context, like they wanted to before. In the new context I just use the goto command to farm the numbers out to separate contexts. This works well. I believe this problem is due to both number being from the same account; which it only want to associate with one context. Regards C From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Tomlinson Sent: 10 March 2005 13:39 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Problem with incoming calls. I have a problem with incoming IAX calls. I have 2 numbers from the same supplier delivered over IAX. I register once with the server, and both calls get to my box, and I get output on the console with both calls. However I cannot get each number to go to separate contexts Please see relevant sections from extensions and iax conf. files Section from IAX.conf: [448700XX] ;incoming 0870 number type=user username=448700XX context=conference trunking=off [448450XX] ;incoming 0845 number type=user username=448450XX context=demo_default trunking=off Section from extensions.conf [demo_default] ;the 0845 number should go here exten = 448450XX,1,Answer exten = ..i have more here. [conference] ;the 0870 number should go here exten = 448700XX,1,Answer exten = ..i have more here. The output on the CLI looks like: NOTICE[1282]: chan_iax2.c:5461 socket_read: Rejected connect attempt from XXX.XXX.X.XXX, request '[EMAIL PROTECTED]' does not exist However if you look above, the 0870 number should go to the [conference] context; not the [demo_default] one.. If I then call the other 0845 number it works: -- Accepting unauthenticated call from XXX.XXX.X.XXX, requested format = 8, actual format = 8 -working If I comment out one of the numbers in the iax.conf, the other one works fine.its just when both are active it doesnt seem to play properly. Does anyone have any ideas? As far as I know I'm not being stupid, but please point it out if I am. Any help much appreciated. Regards, C ___ 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] Xlite dont ring on Asterisk
I have Asterisk configured and can place calls from XLite. But when I call my Asterisk box and try the extension where I'm logged in via my XLite, it doesnt ring and goes immediately to vm. I'm using AMP. 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
[Asterisk-Users] Message Waiting over a IAX trunk
I have Asterisk set up at 2 offices, connected via an IAX trunk. My delema is one person is always moving between offices. I have the dial plan set up to ring phones at both offices but his voicemail box is at office A. His phone at office A has the message indicator, however, he wants to also have the message indicator at office B. Has anyone figured out a way to set the phone registered at Office B to pick up the message waiting indicator from Office A voicemail over the trunk? Martin ___ 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] Sipura 841 Headset microphone volume?
We're setting up some Sipura 841 phones and they're working pretty well, but the microphone volume on the headset (not the handset) is too loud with our Plantronics headsets. Is there some way to turn down the amplification on the headset mic? The microphones are picking up the sound of someone walking on the floor across the room and every little movement or shuffle of the user. I found places to control the output volume of handset, headset, speaker, and ring but I haven't found anything for input control. Can anyone suggest anything? 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] Re: OT: Best DB
You snipped out my paragraph where I mentioned tuning the DB itself, queries, and schema. I have no doubt that 8.x is faster than 7.x, but I did not find any reports from people claiming a 10X performance boost. I didn't look hard, but I did look. I'll look into installing 8.x and see if we can rerun the tests. a fast query is not always enough, if this is your case, then go for mysql. is you look for full featured db, strong performance (may be you can see a mysql query is faster some times, but with postgres you get acceptable response time on every query, no matter how complex it is), great support forums, true open source (free), and you have talented SQL programmers (you need to use more than simple sum,left join, case statements) you must go with postgres, they feel more comfortable with their great PLPG/SQL language, not the limited functions of mysql (read my previus post on this thread for details). --- Miguel ___ 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: Best DB
On Thu, 2005-03-10 at 08:57 -0600, Jay Milk wrote: IB/FB stores the DB in one file, but the file can span multiple drives if needed. However, you can't select which table goes into which file. Personally, I don't think that's very feasible, nor is it required -- if a table is accessed often enough to be mission critical, large parts of it will reside in memory due to caching anyway. Maybe I work in an odd environment where writes(updates and inserts) are probably equal to or more than the reads. Caching isn't real helpful at making the data get to disk faster. Caching helps for reads only. I'll admit I haven't had to use this feature yet, but I see where some people could really need it. -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Sent: Thursday, March 10, 2005 1:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] OT: Best DB If it stores the entire DB in 1 file, it can not scale as well as other DBs. Postgres 8 supports splitting a single DB up so you can put portions of it on different media if needed. If you have to tune for absolute speed, you can purchase one of the solid state drives for the tables that need that kind of speed while using much less expensive harddrives for the rest of the DB. While I do not remember mysql supporting it this directly, I think I remember the file structure being not to difficult to figure out and split and symlink back together if need be. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] OH323 - compilation error (another user, another error)
Hi, pwlib 1.6.6 downloaded ./configure make it as written The same with openh323-1.13.5 Downloaded patched make ./configure make it as written Then with asterisk-oh323-0.7.1 Downloaded (I used u file there to patch openh323) Made some changes in the Makefile to adjust directories Then 'make' I got an error in chan_oh323 : asterisk/channel_pvt.h No such file ... That file (chan_oh323) makes a lot of *.h includes. I found most under /usr/src/asterisk/include/asterisk. Only the 'channel_pvt.h' is missing. I updated via cvs rather recently. I did today a cvs again. but the 'missing file' did not came. And I could not find it in another directory. Regards, Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail [EMAIL PROTECTED] ___ 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: OT: Best DB
Walt Reed [EMAIL PROTECTED] writes: You snipped out my paragraph where I mentioned tuning the DB itself, queries, and schema. Yeah, I see now that it looked as if I were implying that you hadn't. My little list wasn't directed at you as such, I just used your posting as a hook for pointing out to people the importance of tuning. Sorry about the misunderstanding! -tih -- Don't ascribe to stupidity what can be adequately explained by ignorance. ___ 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] Ports/Protocals to Open in Firewall
I am just getting started using Asterisk and would like to know what ports I need to open in my firewall for incoming and outgoing calls. I am running a Cisco Pix 506 and I am having problems using Xlite to make calls through Asterisk = Broadvoice and I think this maybe due to not having the proper protocols passed since I can use X-lite on its own ok from home behind my Linksys Router (no Xlite). Thanks, Scott ___ 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
