Re: [Asterisk-Users] Incoming MSN via ZapHFC - to SIP
Try deleting the line
pritrustusercid=yes
in zapata.conf
maxx
- Original Message -
From: Bastian Schern [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, August 20, 2004 8:34 PM
Subject: [Asterisk-Users] Incoming MSN via ZapHFC - to SIP
Hi there,
I've got a small problem with the zaphfc channel. No MSN of an any
incoming call which comes trough the ISDN card (Acer ISDN, with HFC
chipset and zaphfc driver) which will be forwarded to the SIP-Phone will
be displayed. Always it will be shown asterisk an the Display.
--- snip (zapata.conf) ---
[channels]
language=de
switchtype = euroisdn
signalling = bri_cpe_ptmp
pridialplan=local
prilocaldialplan=local
pritrustusercid = yes
echocancel=yes
immediate=no
group = 1
context=default
channel = 1-2
--- snap ---
--- snip (extensions.conf) ---
[general]
static=yes
writeprotect=yes
[globals]
BASTIAN=SIP/16
[macro-callwithmsn]
exten = s,1,SetCallerID(${ARG2})
exten = s,2,SetCIDName(${ARG3})
exten = s,3,Dial(Zap/g1/${ARG1},60,Ttr)
exten = s,104,Playtones(busy);
exten = s,105,Busy
[default]
exten = 96,1,SetCIDNum(${CALLERIDNUM})
exten = 96,2,Dial(SIP/16)
exten = _0.,1,Macro(callwithmsn,${EXTEN:1},61,Bastian)
exten = _XX,1,Dial(SIP/${EXTEN})
--- snap ---
It would be very nice if somebody can help me.
Regards
Bastian
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Re: [Asterisk-Users] TE410P - ZT_CHANCONFIG failed
Hi, you can also load the wct4xxp driver with modprobe wct4xxp t1e1override=1 Does the same job, remotely. Kind regards, Martin List-Petersen On Fri, 2004-08-20 at 22:11, Arnaud Pignard wrote: Ok, i haven't get in hand the card and make remote hardware install. It's certainely the problem. Thanks ! At 22:49 20/08/2004, you wrote: On Fri, 20 Aug 2004, Arnaud Pignard wrote: I try setup a TE410P. Already setup E100P without problem. I also check sample zaptel.conf config in mailing list and seems my config is ok. However when i modprobe wct4xxp, here is error output : ZT_CHANCONFIG failed on channel 97: No such device or address (6) FATAL: Error running install command for wct4xxp Have you configured the spans for E1 signalling? It sound like you have them set for T1 signalling (24*4=96). You should check the jumpers on the card. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hangup question
Hi, I realize that this topic has been hashed and rehashed a lot of times on this list, but none of the information I was able to find quite fits my case. I seem to have disconnect supervision, namely the light on the handset turns off for ~2 seconds when the other side hangs up, and I have fxs_ks signalling set up on the X100P (clone... it's an IA92) card. Here are a few scenarios of what can happen: (a) SIP - * - PSTN; PSTN hangs up, * doesn't notice (b) PSTN - * - SIP or other extension; PSTN hangs up, * notices (c) SIP - * - PSTN; SIP hangs up, PSTN (my cell phone in this case) does not hang up for another almost exactly 10 seconds. Does anyone have any suggestions as to what may be going on? (I've also tried IAX1 and IAX2, they behave same as SIP as far as I can tell) Also, when SIP hangs up, * lists something along the lines of Hungup 'Zap/1-1', but when I make normal calls to my cell phone and hang up, the cell phone hangs up immediately. Suggestions very much appreciated. Thanks, Ilia Mirkin [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] just-added second X100P
I just added a second X100P card to my * server, altough it seems to be working * seems to be ignoring it: zaptel.conf: - fxsks=1-2 loadzone=us defaultzone=us zapata.conf: -- context=inbound-analog signalling=fxs_ks group=1 channel = 1 channel = 2 I created a couple of test extensions: ; test extensions exten = 4390,1,Dial(Zap/g1/4189) exten = 4390,2,Congestion exten = 4391,1,Dial(Zap/1/4189) exten = 4391,2,Congestion exten = 4392,1,Dial(Zap/2/4189) exten = 4392,2,Congestion 4391 works fine, 4392 doesn't: -- Executing Dial(IAX2/[EMAIL PROTECTED]/2, Zap/2/4189) in new stack Aug 21 02:47:36 NOTICE[426002]: app_dial.c:714 dial_exec: Unable to create chann el of type 'Zap' == Everyone is busy/congested at this time I don't know what's wrong, Zap/2 shows fine in the zap channels list: pbx*CLI zap show channels Chan Extension Context Language MusicOnHold pseudoinbound-analog 1inbound-analog 2inbound-analog Any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura endpoints
I had issues with two sipuras after I upgraded the firmware, after emails and troubleshooting they sent me an RMA. They fixed one and replaced the other. On Fri, 20 Aug 2004 07:43:56 -0500, Matt Schulte [EMAIL PROTECTED] wrote: Anyone have experience with Sipura's? Anyone know if they offer a warranty? Would like opinions on these, good or flame. We bought *one* to test with and it died, can't even get a response from Sipura support. Could anyone recommend another device to replace these? Prefer 1 or 2 port design. Ty :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] telnet and Root
Thanks to all those who responded to my post. Neil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Woods Sent: 21 August 2004 01:05 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] telnet and Root Chris Shaw wrote: If you really want to be able to telnet in as root, locate telnetd.conf or somesuch and it should be in there somewhere as a yes/no. (It is for ssh anyway..) No, not under any distro I'm familiar with... It's under /etc/securetty... You add the tty of the device you want to allow root access to, like pts/0... DON'T DO THIS THOUGH, unless you don't care that your root password will be sent PLAINTEXT over the internet... He may not be telneting to it across the internet. He may only be doing it from his local network. That being said, I like almost everyone else, recommend ssh *and* su, though I'm guilty of logging in as root across the internet with ssh. -Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and AVM FritzBox Fon (Germany)
Hello everyone, I've some stange problems with my AVM FritzBox Fon (http://www.avm.de/de/Presse/Informationen/2004/2004_06_22.php3) and Asterisk. I want to build up an internal SIP Network for testing purposes. I can phone with my Grandstream BT-101 to other BT-101 or SIP-Software, but not to my AVM FritzBox Fon. It looks like the Box connects successfully to Asterisk, but you can't call other internal SIP Accounts. You also can't call to the Box. The Phone on the Box rings, but I you want to start the call you hear nothing. And after a few seconds, the Grandstream responds a 403-Error. Here a snapshot from my sip.conf [general] port = 5060 bindaddr = 0.0.0.0 context = default [10] type=friend username=10 fromuser=10 host=dynamic secret=SECRET canreinvide=no [11] type=friend username=11 fromuser=11 host=dynamic secret=SECRET Here my extensions.conf [general] static=yes writeprotect=no [default] include = 10 include = 11 [10] exten = 10,1,Dial(SIP/10,45) exten = 10,2,Hangup [11] exten = 11,1,Dial(SIP/11,45) exten = 11,2,Hangup I've tried a qualify=yes in the sip.conf, but then the Box will loose connection after a few seconds: Aug 21 13:16:10 NOTICE[98310]: chan_sip.c:7653 sip_poke_noanswer: Peer '10' is now UNREACHABLE! I would be really pleased for any help :) Regards, Marco canreinvide=no ___ Gesendet von Yahoo! Mail - Jetzt mit 100MB Speicher kostenlos - Hier anmelden: http://mail.yahoo.de ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] system reboot often?
I just deployed * on my home system last Sunday. 2x since then the Zap hardware seems to have malfunctioned on some way. One time it would just screech out one FXS, even though it would ring. The other time * would bridge to my FXO but it never got out on the line. I have a new TDM400 with 3 FXS and 1 FXO. Both times I tried unloading the zaptel drivers (which worked) and reloading them, which failed. A reboot of the system brought everything back. Is this common? Are there ways to minimize this? Would a different PCI slot possibly make a difference? Or a different system? Is this just a chronic problem with the Digium hardware? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] system reboot often?
On Sat, 2004-08-21 at 06:42, Michael George wrote: I just deployed * on my home system last Sunday. 2x since then the Zap hardware seems to have malfunctioned on some way. One time it would just screech out one FXS, even though it would ring. The other time * would bridge to my FXO but it never got out on the line. I have a new TDM400 with 3 FXS and 1 FXO. Both times I tried unloading the zaptel drivers (which worked) and reloading them, which failed. A reboot of the system brought everything back. Is this common? Are there ways to minimize this? Would a different PCI slot possibly make a difference? Or a different system? Is this just a chronic problem with the Digium hardware? Most problems with TDM400 cards seem to stem from power supply. That is why there is a power plug on the card to get more power into the card to support the phone signalling. You may have had a small power brown out and it caused your power supply to drop below what the card was happy with. Make sure you have a quality power supply in the computer that has plenty of surplus wattage. Then make sure your machine is on a UPS to make sure the power supply has enough clean power to serve your needs. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] welltech fxo and *
hello everybody , i got a test setup with lanphone 101 connected to * and a welltech fxo 3802 also to * the extensions are configure so that i can dial from the lanphone to the fxo. although once on the FXO and having the dialtone, no of my dtmf dialing is being processed on the FXO. It keeps giving me the dialtone constatly. Everything is configured as outband DTMF (we tried also doing inband but it was doing the same) We tried to call the echo test application to check if the # would close the connection, and it does. So dtmf from lanphone to * seems to work OK. We also tried to configure a extension like dial(SIP/FXO,50,D(0022156765)) it does connect to the fxo; but the dtmf isn't being processed... (keeps giving me the dialtone) Anybody got a similar problem like this, and a prob. solution ? the sip.conf for each unit looks like [home] disallow=all allow=ulaw type=friend secret= username=home host=dynamic context=default nat=yes dtmfmode=rfc2833 -- Best regards, Danny mailto:[EMAIL PROTECTED] belGOnet.com a Euro-pictures division - internet solutions place princesse elisabeth 9/11 - 1030 Brussels - Belgium Tel : +32-(0)2-215.67.65 - Fax : +32-(0)2-215.66.65 domains - hosting - hardware - VoiP - consultancy - backuping CISCO - HP/COMPAQ - SUN - EMC - JUNIPER - IBM - DELL - NORTEL No legal consequences can be derived from the contents of the email neither is belGOnet.com committed to them. The content of this email is exclusively intended for adressee(s) and information purposes. belGOnet.com accepts no liability for any damage resulting from the use and/or acceptation of the content of this email. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] system reboot often?
This doesn't answer your question completely, but I have noticed that inserting and removing the kernal modules doesn't work all that well and that rebooting is a better answer at that point. Have you verified that you are not IRQ sharing? * really doesn't like that, even though other applications are ok with it. - Original Message - From: Michael George [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, August 21, 2004 6:42 AM Subject: [Asterisk-Users] system reboot often? I just deployed * on my home system last Sunday. 2x since then the Zap hardware seems to have malfunctioned on some way. One time it would just screech out one FXS, even though it would ring. The other time * would bridge to my FXO but it never got out on the line. I have a new TDM400 with 3 FXS and 1 FXO. Both times I tried unloading the zaptel drivers (which worked) and reloading them, which failed. A reboot of the system brought everything back. Is this common? Are there ways to minimize this? Would a different PCI slot possibly make a difference? Or a different system? Is this just a chronic problem with the Digium hardware? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel config
Hi, Sorry, in my last mail I wrote wcfxs instead of what I actually used, wcfxo. I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxo, which worked fine. ztcfg is showing two channels configured, but when I start asterisk and do show channels, i see no active channels. zapata.conf has: signalling = fxs_ks context = line1 channel = 1 signalling = fxs_ks context = line1 channel = 2 zaptel.conf has: loadzone=us defaultzone=us fxsks=1-2 extensions.conf has: [line1] exten = s,1,Answer exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,10 exten = s,4,BackGround(demo-congrats) exten = s,5,BackGround(demo-instruct) [line2] exten = s,1,Answer exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,10 exten = s,4,BackGround(demo-congrats) exten = s,5,BackGround(demo-instruct) I have no idea why it's not working would appreciate any help Thanks, Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaptel config
show channels only shows active calls. So if no calls are active, there are no channels active to be displayed. Try 'zap show channels' first. Then 'asterisk -c' and call one of the cards and see what pops up on the screen. It should give you clues as to the problem. Lyle - Original Message - From: Imran Akbar [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, August 21, 2004 9:03 AM Subject: [Asterisk-Users] zaptel config Hi, Sorry, in my last mail I wrote wcfxs instead of what I actually used, wcfxo. I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxo, which worked fine. ztcfg is showing two channels configured, but when I start asterisk and do show channels, i see no active channels. zapata.conf has: signalling = fxs_ks context = line1 channel = 1 signalling = fxs_ks context = line1 channel = 2 zaptel.conf has: loadzone=us defaultzone=us fxsks=1-2 extensions.conf has: [line1] exten = s,1,Answer exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,10 exten = s,4,BackGround(demo-congrats) exten = s,5,BackGround(demo-instruct) [line2] exten = s,1,Answer exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,10 exten = s,4,BackGround(demo-congrats) exten = s,5,BackGround(demo-instruct) I have no idea why it's not working would appreciate any help Thanks, Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] just-added second X100P
Although it shouldn't make a difference, try: channel = 1-2 As well, did you run ztcfg after you installed the new card? I've found sometimes I've had run ztcfg a couple times before Asterisk would kick in and recognize a new card. On Sat, 2004-08-21 at 02:49 -0500, spectro wrote: I just added a second X100P card to my * server, altough it seems to be working * seems to be ignoring it: zaptel.conf: - fxsks=1-2 loadzone=us defaultzone=us zapata.conf: -- context=inbound-analog signalling=fxs_ks group=1 channel = 1 channel = 2 I created a couple of test extensions: ; test extensions exten = 4390,1,Dial(Zap/g1/4189) exten = 4390,2,Congestion exten = 4391,1,Dial(Zap/1/4189) exten = 4391,2,Congestion exten = 4392,1,Dial(Zap/2/4189) exten = 4392,2,Congestion 4391 works fine, 4392 doesn't: -- Executing Dial(IAX2/[EMAIL PROTECTED]/2, Zap/2/4189) in new stack Aug 21 02:47:36 NOTICE[426002]: app_dial.c:714 dial_exec: Unable to create chann el of type 'Zap' == Everyone is busy/congested at this time I don't know what's wrong, Zap/2 shows fine in the zap channels list: pbx*CLI zap show channels Chan Extension Context Language MusicOnHold pseudoinbound-analog 1inbound-analog 2inbound-analog Any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Finding operator from ISDN signalling?
On Fri, 20 Aug 2004, Roy Sigurd Karlsbakk wrote: Is it possible to find out what source/destination operator you're connected to from the ISDN signalling? Number lookups no-longer work since number porting begun, and access to the national database for these numbers cost too much :( That information is, unfortunately, not included in signalling. In Sweden you can get the information for free (restricted number of queries) by a web service at Swedish Number Portability Administrative Center. I do not know if there is somethink similar in Norway. Regards, Tobias Jönsson, Lund SE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] residential sip phone
Chris Shaw wrote: These phones can be found for as low as $65 USD and have EXCELLENT voice quality and can even work behind NAT... I disagree. I think the Grandstreams sound like crap (at least on the remote end, I've never used one. Just heard people on one.). Lots of bad echo. -- Daniel Jimenez djimenez[at]pobox[dot]com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] autocreatepeer and sip peer options
Hi all, quick question...i am using autocreatepeer to get asterisk to work with SER without having to specify each UA in sip.conf and in ser separately. 2 questions: 1. obviously this is not very secure. assuming i block incoming requests on the port asterisk is running SIP on (excluding requests from the SER, of course) does this adequately protect the server from unauthorized users or is there something else to do? 2. according to the wiki the autocreatepeer creates peers based on the global variables. some variables, like dtmfmode, for example, are listed as belonging to individual peers. if i set dtmfmode, or qualify, or any of the others listed as individual variables, in [general] will the autocreatepeer use them? I suppose i could write a script to automatically generate peers for asterisk from SER's DB, (along the lines of the current retrieve_sip_conf_from_mysql.pl but having duplicate SIP client entries seems kind of inelegant. any help is appreciated, thanks- yair mail2web - Check your email from the web at http://mail2web.com/ . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to minimally configure modules.conf loading?
I'm trying to somewhat reduce the security risk of Asterisk, by loading less modules. In my installation I use SIP and IAX2 for incoming calls, and that's it. No voicemail, no call parking, it just plays back voice clips. I can remove /etc/asterisk/modules.conf modules one by one: [modules] autoload=yes noload = pbx_gtkconsole.so noload = pbx_kdeconsole.so noload = app_intercom.so ;obsolete noload = chan_alsa.so noload = chan_oss.so noload = chan_skinny.so noload = chan_phone.so noload = app_voicemail.so noload = chan_zap.so noload = app_meetme.so But I've not made it very far building up just what I need: [modules] autoload=yes load = res_crypto.so load = res_features.so load = chan_iax2.so load = chan_sip.so load = codec_gsm.so load = codec_ulaw.so I get hard to track down symbol errors like: [chan_iax2.so]Aug 21 10:37:02 WARNING[16384]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_iax2.so: undefined symbol: ast_check_signature Aug 21 10:37:02 WARNING[16384]: loader.c:374 load_modules: Loading module chan_iax2.so failed! [chan_iax2.so]Aug 21 10:38:12 WARNING[16384]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_iax2.so: undefined symbol: ast_moh_stop Does anyone know a better way to do this? Grep'ing the soruce by trial and error is not working. Modules have to be in just the right order. -Bryce ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk in india
Hi Navit, My company is developing asterisk compatible HW SW. If you would like to contact me, we might be able to work on your wishlist. Eran Gal - Xorcom [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Navnit Chachan Sent: Tuesday, August 17, 2004 8:22 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] asterisk in india Hi, The company i work for has been active with asterisk for more than a year and have done plenty of * installations . Their VOIP team is based in delhi. Navnit - Original Message - From: Kannaiyan Natesan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, August 14, 2004 2:07 AM Subject: Re: [Asterisk-Users] asterisk in india You can use it for interconnection with the local corporate office. Internet --- Asterisk -- TE405 -- (E1/T1) Channel Banks FXS Ports to company employees Which is legally allowed in INDIA. Not sure of any resellers in INDIA at this moment. -Kannaiyan - Original Message - From: Asterisk . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 13, 2004 3:07 PM Subject: Re: [Asterisk-Users] asterisk in india Hi, --- Vikram Rangnekar [EMAIL PROTECTED] wrote: Does anyone know if the E1 cards that digium sells work in India. Also are there any distributers for those cards in India. By E1 cards I mean E100P, TE410P or TE405P DoT rules dont permit VoIP-PSTN termination is in India, it is illegal. However, you can make calls from PC to PC, and PC to Phone where PC in India and Phone abroad. AFAIK, there are no resellers for Digium cards here. -- regards Vikram (http://www.vicramresearch.com) Regards, Girish Gopinath __ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multi-bitrate codecs
Simone Ricci [EMAIL PROTECTED] wrote: Anyone knows if there's a way to select the bitrate of those codecs supporting multiple bitrates (eg. g.726)? I've tried searching and googling a lot, but without useful results... Asterisk's G.726 codec only supports the 32-bit mode. You can't select 40-bit nor any of the others, and it can't negotiate for itself. I'm sure someone will leap in to correct me if I'm wrong. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] residential sip phone
on 8/21/04 13:37, Daniel Jimenez at [EMAIL PROTECTED] wrote: I disagree. I think the Grandstreams sound like crap (at least on the remote end, I've never used one. Just heard people on one.). Lots of bad echo. I would tend to agree with Daniel. and the workmanship on them are terrible. We would like to sell the ones we have...any takers? .shawn. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zultys 4x4 or 4x5
Hi All, A friend has asked me to get a new Zultys 4x5 working with his * server. I've been over the web gui but don't see where all the registration info gets entered in the phone. Is there someone on-list who has Zultys phones and can advise me as to their setup? The manufacturers docs are useless. Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 Lawyers, guns and money can't get me out of this. - Warren Zevon ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Testing null values: ast_yyerror(): syntax error
Dnia roda, 18 sierpnia 2004 16:56, Walt Reed napisa:
OK, I'm going nuts here trying to correctly identify null values,
specifically when callerID info is not available.
FYI, I'm running Asterisk CVS-HEAD-08/17/04-13:08:53, and Bison 1.875a
(debian Sid).
A snippit of my dialplan looks like this:
exten = s,1,SetCIDNum(${CALLERIDNUM})
exten = s,2,NoOp,${CALLERID}
exten = s,3,DBGet(temp=idiot/${CALLERIDNUM}) ; Is the person calling an
idiot? exten = s,4,Goto(s,2001) ; Yep, he's an idiot.
Don't know whether there's a more elegant solution, but maybe try putting
values like x${CALLERIDNUM} into the DB? that way if you get a null CID, it
should match against x.
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[Asterisk-Users] Asterisk and software Raid
I've got a Dual Xeon system Digium Quad T1 and an Idle T1 circuit that I can experiment with.I've been wanting to use Redhat with software Raid 1 on an Asterisk server. Has anyone had any experience with software raid and Asterisk? Also, if the software raid doesn't play, any recommendations for a hardware based IDE Raid controller and suggestions on best practices for setting up the disk partitions (2 X 40 GB. Maxtors) for a mirrored environment would be appreciated. Thanks all
Re: [Asterisk-Users] Asterisk and software Raid
I used software raids and want to get away from them. I really really like Microlite's BackupEdge tape backup software. BackupEdge does not work with a software raid, only a hardware raid. The second kicker was that the Promise (or any other) hardware IDE raids are considered software raid to the Kernel. And are NOT supported with the new 2.6.x Kernels... My experience with IDEin raid arraysis less than stellar and will be trashing them as I rebuild servers. I have had several instances where one drive fails and the entirearray falls overas the kernel struggles to recover from the loss of a drive or the error messages. I have seen this with Linux generatedarrays and with the Promise IDE raid cards. Besides the performance of the Promise parallel IDE raid sucks big time. With the excellent tape software above, I don't think I really need them anyway. BackupEdge will generate a bootable CD-Rom that will install your last backup on bare metal after a major malfunction. Lyle - Original Message - From: Ed Devine To: [EMAIL PROTECTED] Sent: Saturday, August 21, 2004 1:47 PM Subject: [Asterisk-Users] Asterisk and software Raid I've got a Dual Xeon system Digium Quad T1 and an Idle T1 circuit that I can experiment with.I've been wanting to use Redhat with software Raid 1 on an Asterisk server. Has anyone had any experience with software raid and Asterisk? Also, if the software raid doesn't play, any recommendations for a hardware based IDE Raid controller and suggestions on best practices for setting up the disk partitions (2 X 40 GB. Maxtors) for a mirrored environment would be appreciated. Thanks all
[Asterisk-Users] cmd Monitor creating sound notification on channel
started messing around with the Monitor cmd for call recording but noticed that the cmd was 'injecting' a small noise every 8 sec or so. looked through the wiki for a flag setting but the 'm' for using soxmix post call is the only one noted. trying to build a call logger that just sits next to another switch and does the recording but if it gives an indication that it is recording and I can't change that I will have difficulty selling it to the powers that be. Does anyone know of a flag setting that makes 'Monitor' silent? Jason Kawakami ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and software Raid
Ed Devine wrote: I've got a Dual Xeon system Digium Quad T1 and an Idle T1 circuit that I can experiment with. I've been wanting to use Redhat with software Raid 1 on an Asterisk server. Has anyone had any experience with software raid and Asterisk? Also, if the software raid doesn't play, any recommendations for a hardware based IDE Raid controller and suggestions on best practices for setting up the disk partitions (2 X 40 GB. Maxtors) for a mirrored environment would be appreciated. Thanks all I've had no luck with software RAID in Linux. Using RAID1, recovering a drive seems to peg the processor, and the recovery lasts for over 16 hours for 160GB disks. I can't even imagine what it is doing during that period. I played with the /proc values, but I wasn't able to reduce the CPU utilization. Needless to say, with 90-100% CPU utilization, voice quality goes down the toilet--the system was totally unusable until the recovery was complete. I've gone to the 3Ware 7006-2 ($100). This card has two IDE ports (other cards have four, eight, and 12). To the system it appears as a SCSI controller with one drive. Drive recovery is handled on the card. I have an eight port card with 200MB drives on my database machine (RAID5), and it works quite well. 1.4TB of disk. I've lost a drive and the machine continues to operate. I'm sorry to hear about the Promise chip because that is what is on the newer Tyan mother boards. For a 1U or 2U cabinet, I don't have the luxury of a bunch of PCI slots (the two slots are occupied by T405P cards), and I need to use the on-board raid. Does anyone else have any experience with this. Cheers, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and software Raid
On Sat, 21 Aug 2004, Ed Devine wrote: I've got a Dual Xeon system Digium Quad T1 and an Idle T1 circuit that I can experiment with. I've been wanting to use Redhat with software Raid 1 on an Asterisk server. Has anyone had any experience with software raid and Asterisk? Also, if the software raid doesn't play, any recommendations for a hardware based IDE Raid controller and suggestions on best practices for setting up the disk partitions (2 X 40 GB. Maxtors) for a mirrored environment would be appreciated. I have a Dell, 1 Ghz PIII 1U rackmount server, w/ a T100P (PRI) running Tao Linux (RedHat Enterprise 3.0) on 2 40 gig drives in RAID-1. I've had as many as 20 calls on the box at one time, and never noticed an issue. Of course, 90% of our calls are pass-through Ulaw, and although we have g729 licenses it doesn't get used very much. Never had a problem with it so far. However, if you are looking for an IDE Hardware Raid solution, I have tried everything on the market, and I can honestly say that for price, compatibility and reliability, the 3Ware cards are the best bet. This is the only card that we spec for clients, as the Promise and Adaptec stuff is garbage. Incidentally, if you want a good deal on a Promise Fast-Track SX 6000, I have a few laying aroung! ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and software Raid
On Sat, 21 Aug 2004, Lyle Giese wrote: I used software raids and want to get away from them. I really really like Microlite's BackupEdge tape backup software. BackupEdge does not work with a software raid, only a hardware raid. The second kicker was that the Promise (or any other) hardware IDE raids are considered software raid to the Kernel. And are NOT supported with the new 2.6.x Kernels... This is NOT generally true. Many of the embedded Promise controllers, built into the motherboards, are detected as separate devices. My experience with IDE in raid arrays is less than stellar and will be trashing them as I rebuild servers. I have had several instances where one drive fails and the entire array falls over as the kernel struggles to recover from the loss of a drive or the error messages. I have seen this with Linux generated arrays and with the Promise IDE raid cards. Besides the performance of the Promise parallel IDE raid sucks big time. Well, the Promise cards suck in general. That's not Linux's fault! :) With the excellent tape software above, I don't think I really need them anyway. BackupEdge will generate a bootable CD-Rom that will install your last backup on bare metal after a major malfunction. Here is another vote for MicroLit's Backup Edge. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] residential sip phone
Depending on the application the Grandstream is decent but for prolonged use I've found it's better to not pinch the pennies and go with something a bit more expensive but with less problems. For a simple SOHO deployment I'm just putting a Sipura SPA 3000 on their cordless base station (Panasonic). Hopefully given time a sub $100 phone does come up without the shortcomings of those before it but until then I'd be hard pressed as to what phone to deploy. -- William On Sat, 21 Aug 2004 14:04:44 -0400, programming dept [EMAIL PROTECTED] wrote: on 8/21/04 13:37, Daniel Jimenez at [EMAIL PROTECTED] wrote: I disagree. I think the Grandstreams sound like crap (at least on the remote end, I've never used one. Just heard people on one.). Lots of bad echo. I would tend to agree with Daniel. and the workmanship on them are terrible. We would like to sell the ones we have...any takers? ..shawn. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and software Raid
Lyle Giese [EMAIL PROTECTED] writes: I used software raids and want to get away from them. I really really like Microlite's BackupEdge tape backup software. BackupEdge does not work with a software raid, only a hardware raid. I have worked with software raid for several years, and to date have not had any problem with that. Of course the machine is more prone to reboots in case of drive failure (the hardware is not designed to recover from such events), but it usually just reboots, array in degraded mode, data intact, and ready to assume it's duties. The second kicker was that the Promise (or any other) hardware IDE raids are considered software raid to the Kernel. And are NOT supported with the new 2.6.x Kernels... Promise and such are really low end raids adapters. For good ide hardware raid, stick with 3ware. The price tag is much higer than promise, but these are real professionnal raid adapters. They are seen as scsi devices by linux, the driver is completely GPL, and has been part of the official kernel tree for years. With these, it takes really niche situations to ever need scsi raid again. My experience with IDE in raid arrays is less than stellar and will be trashing them as I rebuild servers. I have had several instances where one drive fails and the entire array falls over as the kernel struggles to recover from the loss of a drive or the error messages. I have seen this with Linux generated arrays and with the Promise IDE raid cards. Besides the performance of the Promise parallel IDE raid sucks big time. Promise are not real professionnal grade products, and software raid can do nothing against hardware shortcomings. The only way to be protected is with a real pro raid controler. With the excellent tape software above, I don't think I really need them anyway. BackupEdge will generate a bootable CD-Rom that will install your last backup on bare metal after a major malfunction. Backups are not subsitutes for raid, and vice versa. couic Back to *, I have only one * installation on software raid, so I can't really make statistics, but it has no problem. Bye, -- Rémi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can I place calls to arbitrary sip uri address from windows messenger 5 ?
Hi, I've sucessfully registered windows messenger to Asterisk and can talk to other extensions cause they appear as being online. But can I place a call to arbitrary SIP address (if I enter it to contacts it appears as offline and don't get possibility to start voice or video conversation) ? Are there any other free softphones for windows that would support video also (sip or iax) ? What are other practical experiences using WM and Asterisk ? Thanks in advance, regards, Robert. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Number and name for SIP extension at the same time ?
Hi, I'd like to have local extensions accessible through SIP uri (like [EMAIL PROTECTED]), but at the same time for convenince to be also extension with number (like 100) for more convenient dialing thought softphones that support only numeric keys. Can this be done ? Since I'm newbie, I'd really appreciate small example... Thanks in advance, regards, Robert. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ActXPhone Active X control link is dead - has anyone cached files ?
Hi, I'm interested in ActXPhone active x web control (mentioned on voip-info.org) but link is dead. Has anyone cached files and is willing to share them ? Thanks, Robert. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] determining what number was dialed?
Here's our setup:
8 analog lines all terminate in an Adtran TA750 channel bank we
bought. From there a crossover cable connects to the T100P card in our
Asterisk server.
(sorry, I'm more of a perl/java guy who got thrown into this, my phone
knowledge is pretty weak). I'm not too sure what signalling info
you're looking for. The settings in the channel bank are defaults, esf
framing and b8zs coding.
We temporarily had regular phones connected to a few of the analog
lines and the caller ID information did come through. Are there
questions I should ask my provider that will tell me what kind of
signalling we use?
In the meantime, I'll see what ${DNID} can do ...
Thanks,
-paul
On Fri, 20 Aug 2004 16:08:53 -0500, Steven Critchfield
[EMAIL PROTECTED] wrote:
On Fri, 2004-08-20 at 15:47, Chris Shaw wrote:
True, very true... If it's PRI then you will get DNIS/DNID from the
D-Channel...
If they're doing anything other than PRI though, like a regular T1 into a
channel bank (or into a TE100P or TE40xP) , this would work... Lines would
be assigned to a specific channel and they could be separated out with
contexts... Not as pretty as using DNIS but it would work...
Even a regular T1 can have DID and therefore again, the line is not
relavent. We had a channelized T1 with Em wink and the lines where
not assigned a number, we decoded it as part of call setup via dtmf. On
those lines I know we didn't receive the 1800 number as it was forwarded
to our DID number and we just received the last 4 digits of the DID.
So we are back to waiting to hear from the PSTN connectivity and
signalling before we can continue giving advice.
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, August 20, 2004 12:33 PM
Subject: Re: [Asterisk-Users] determining what number was dialed?
On Fri, 2004-08-20 at 14:28, Paul Concepcion wrote:
Hey all,
I've setup * to serve the needs of our small helpdesk and I'm
looking
to expand. We're planning on doing support for different companies,
each one identified by a different 1-800 number that terminates at
our
PBX. What I would like to know is: is there a variable I can read to
determine what number any given caller dialed? I'd like to be able
to
separate calls based on who called 1-800-777- and who dialed
1-800-555-, for example.
Yes, but it depends on what type of telephony signalling you are using
as to whether or not you can get that information.
Tells us about your PSTN connection.
--
Steven Critchfield [EMAIL PROTECTED]
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[Asterisk-Users] Queue Monitor
I understand that putting monitor-format in the queues.conf file will
start monitor recording of an active queue call. Is there a way to
automatically do the post call processing like the 'm' option like when
specifying the use of the monitor command?
Kevin
Monitor(wav,${CALLFILENAME},m)
flags: If flags contains the letter m, then when recording finishes,
Asterisk will execute a unix program to combine the two sound files into
a single sound file. By default, Asterisk will execute soxmix and then
delete the original two sound files.
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Re: [Asterisk-Users] determining what number was dialed?
Paul Concepcion wrote: 8 analog lines all terminate in an Adtran TA750 channel bank we bought. From there a crossover cable connects to the T100P card in our Asterisk server. With this arrangement you cannot receive DNIS. There is no way for the telco to transmit the dialed number to you. We temporarily had regular phones connected to a few of the analog lines and the caller ID information did come through. Are there questions I should ask my provider that will tell me what kind of signalling we use? Caller ID is something else entirely; it is _not_ the number that was dialed, it is (presumably) the number of the caller. If you want to receive the number that the caller dialed (so you can route the call based on that information), you have only two choices: a T-1 from your telco with two-way DID trunks on it, or a PRI from your telco. Asterisk does not support any other means of receiving the dialed number (and there is only one other means, analog DID trunks). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and software Raid
On Saturday 21 August 2004 14:47, Ed Devine wrote: Has anyone had any experience with software raid and Asterisk? Also, if the software raid doesn't play, any recommendations for a hardware based IDE Raid controller and suggestions on best practices for setting up the disk partitions (2 X 40 GB. Maxtors) for a mirrored environment would be appreciated. Supermicro server system using software RAID1 on two 9.1G UW2 drives. No issues thus far. ~3 months running. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and software Raid
On Saturday 21 August 2004 15:17, Lyle Giese wrote: My experience with IDE in raid arrays is less than stellar and will be trashing them as I rebuild servers. I have had several instances where one drive fails and the entire array falls over as the kernel struggles to recover from the loss of a drive or the error messages. I have seen this with Linux generated arrays and with the Promise IDE raid cards. Besides the performance of the Promise parallel IDE raid sucks big time. Odd, I have absolutely *zero* issues with Promise PATA cards... I use strictly software RAID on both SCSI and IDE on Linux 2.4. Never had issues with the kernel failing due to I/O load on rebuild or dealing with failed drives. Note: You can easily throttle the I/O bandwidth used for rebuilding through /proc. I've never had to do it though. Now mind you all I do is software RAID1. I don't do RAID5. I typically buy drives in pairs and then use LVM to give me a big blob of storage and partition it up as I see fit with logical volumes. The largest (# of drives) array I have is an 8-drive array, with 6 in pairs and ganged together for about 300G and then a separate RAID0 on a pair of old IBM DeathStar drives for my temporary data area for MythTv. This is all on a cheapass Pentium3 system. No issues even when running in degraded mode. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] determining what number was dialed?
Thanks for the information. Sorry about tossing that Caller ID tidbit, a separate train of thought appeared while I was typing that last reply. On Sat, 21 Aug 2004 14:23:46 -0700, Kevin P. Fleming [EMAIL PROTECTED] wrote: Paul Concepcion wrote: 8 analog lines all terminate in an Adtran TA750 channel bank we bought. From there a crossover cable connects to the T100P card in our Asterisk server. With this arrangement you cannot receive DNIS. There is no way for the telco to transmit the dialed number to you. We temporarily had regular phones connected to a few of the analog lines and the caller ID information did come through. Are there questions I should ask my provider that will tell me what kind of signalling we use? Caller ID is something else entirely; it is _not_ the number that was dialed, it is (presumably) the number of the caller. If you want to receive the number that the caller dialed (so you can route the call based on that information), you have only two choices: a T-1 from your telco with two-way DID trunks on it, or a PRI from your telco. Asterisk does not support any other means of receiving the dialed number (and there is only one other means, analog DID trunks). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] determining what number was dialed?
well, that's our setup (8 analog lines - channel bank - t100P), so it looks like DNIS is out of the question. We do have 8 phone numbers though. Could we have a 1-800 number direct to each of those, then do what you suggested with contexts? What would happen if two people dialed 1-800-a if 1-800-a was pointed to just one phone number? On Fri, 20 Aug 2004 13:47:30 -0700, Chris Shaw [EMAIL PROTECTED] wrote: True, very true... If it's PRI then you will get DNIS/DNID from the D-Channel... If they're doing anything other than PRI though, like a regular T1 into a channel bank (or into a TE100P or TE40xP) , this would work... Lines would be assigned to a specific channel and they could be separated out with contexts... Not as pretty as using DNIS but it would work... -Chris - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 20, 2004 1:29 PM Subject: Re: [Asterisk-Users] determining what number was dialed? On Fri, 2004-08-20 at 15:23, Chris Shaw wrote: Why not use separate contexts for these lines in zapata.conf? Seems way simpler to me... http://www.voip-info.org/wiki-Asterisk+config+zapata.conf Who said they are seperate lines. 1-800 numbers can just be redirects to other lines. In that case you have to have a different signaling method to determine 1800-a on line b is different from 1800-c on line b. In a PRI circuit, the lines are just channels, all signaling data is run over the D channel and any line can be any number routed to that circuit. - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 20, 2004 12:33 PM Subject: Re: [Asterisk-Users] determining what number was dialed? On Fri, 2004-08-20 at 14:28, Paul Concepcion wrote: Hey all, I've setup * to serve the needs of our small helpdesk and I'm looking to expand. We're planning on doing support for different companies, each one identified by a different 1-800 number that terminates at our PBX. What I would like to know is: is there a variable I can read to determine what number any given caller dialed? I'd like to be able to separate calls based on who called 1-800-777- and who dialed 1-800-555-, for example. Yes, but it depends on what type of telephony signalling you are using as to whether or not you can get that information. Tells us about your PSTN connection. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and software Raid
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Saturday 21 August 2004 05:22 pm, Andrew Kohlsmith wrote: On Saturday 21 August 2004 14:47, Ed Devine wrote: Has anyone had any experience with software raid and Asterisk? Also, if the software raid doesn't play, any recommendations for a hardware based IDE Raid controller and suggestions on best practices for setting up the disk partitions (2 X 40 GB. Maxtors) for a mirrored environment would be appreciated. Supermicro server system using software RAID1 on two 9.1G UW2 drives. No issues thus far. ~3 months running. Yeah, I've run s/w radi for long time and never really had any problems. I just replace the drive when one goes bad. Since I always put a spare it works seemlessly. I just have never done it with Asterisk. I don't think I would use s/w raid with it either. I don't want my cpu's to be busy with drives if it's not needed. SCSI then becomes a good choice too. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBJ8C8ljK16xgETzkRArLWAKDPm/nV75BEnYGhpZXZi160vJ9jGwCfVqD5 fn+85yuBLyAdCIDnyFjeVgM= =2txp -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Inbound IAX2 calls has no music on hold
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Friday 20 August 2004 02:25 pm, Eric Wieling wrote: On Fri, 2004-08-20 at 12:52, Steve Szmidt wrote: Does ANYONE have music on hold working across IAX2? Google does not return anything on the subject. Except I did see on the release notes for 0.7.0 Better support for MOH in IAX2 Yes. It works fine with no special config required. Well, I'm glad it can work. I've never heard any MOH when the call comes in on IAX2. It would be nice to know what the condition is that stops it on IAX2 alone. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBJ8HLljK16xgETzkRAi7PAJwIl58TM6nuqLR2EXiWivhGuw4YUwCg4X0G y0jlQqIfWhcWQt0rsFILMxU= =trnX -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] need help with zaptel.
I cant get zaptel to make. I get this error: make: *** [linux26] Error 1. The previous line is: Link /usr/src/linux-2.6 to your kernel sources first! I am running Fedora Core2 Asterisk compiles fine. I am using my SIP phones. I would like to get my TDM400p working.
Re: [Asterisk-Users] Asterisk and software Raid
I've got a nearly identical system running on Slackware 9.1 with a Megaraid controller. Works better than expected! Niles I've got a Dual Xeon system Digium Quad T1 and an Idle T1 circuit that I can experiment with. I've been wanting to use Redhat with software Raid 1 on an Asterisk server. Has anyone had any experience with software raid and Asterisk? Also, if the software raid doesn't play, any recommendations for a hardware based IDE Raid controller and suggestions on best practices for setting up the disk partitions (2 X 40 GB. Maxtors) for a mirrored environment would be appreciated. Thanks all ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] need help with zaptel.
Look in /usr/src. You should see a directory something similar to linux-2.6.1-1[as an example]. If you DON'T have a directory (or link to a directory) named linux-2.6, you should create one using the 'ln' command. In the case mentioned above, the command would be: ln -s /usr/src/linux-2.6.1-1 /usr/src/linux-2.6 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edward HuittSent: Saturday, August 21, 2004 5:42 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] need help with zaptel. I cant get zaptel to make. I get this error: make: *** [linux26] Error 1. The previous line is: Link /usr/src/linux-2.6 to your kernel sources first! I am running Fedora Core2 Asterisk compiles fine. I am using my SIP phones. I would like to get my TDM400p working.
[Asterisk-Users] Cisco IP Phone- disjoin conference
Hi all, Does anyone know if it's possible to continue talking to one member of an initiated conference call on Cisco 79xx ? In other words- disconnect one of the parties. Thanks. Assaf
[Asterisk-Users] IAX2 DTMF not recognized - Bug report - Help sought
I have working SIP numbers with broadvoice, and just added a DID from
http://connect.voicepulse.com/ . The calls answer, but DTMF is not
recognized.
With iax2 debug active pressing DTMF does nothing. Zilch. Zero.
A friend tried a different IAX2 connection, and got the same results.
I see the following in the archives:
On Fri, 2004-04-09 at 10:12, Robert Jackson wrote:
Hey all,
I am dialing a DID through VoicePulse Connect. The number is
answered by a main menu type of IVR. The configuration is as specified
in both the wiki and VoicePulses documentation. The call comes through
without a problem, but when the caller enter any keys they are either
not recieved by * or they are ignored. With SIP I would typically put a
dtmfmode= line under the peer and everything works great, but I am not
sure how to attack this. I found a few items referring to the same
issue in the list, but I didn't find any answers. If this is a bug I
will create a report on the bugtracker, but I would rather make sure
that I am not just completely dense and not seeing the easy answer. I'm
trying to replicate the issue with NuFone.
CVS from 2004-04-04 stable branch.
JC wrote on Wed, 28 Jan 2004 19:47:41 -0500
Hello all, I am using voicepulse DID's to receive calls via IAX to and =
asterisk IVR dial plan I have put together. The problem is after 3-5mins =
the system cant pickup the DTMF tones I am sending... I have tried =
different telephones... It just repeats menu options over and over I =
have to call back and then it works again for another few mins...
Any ideas... iax.conf? issue?
Thanks,
J.C.
Chris,
Thank you for contacting VoicePulse.
Our engineers are aware of the DTMF problem and are working to have it
resolved as quickly as possible.
Please reply directly to this email if we can provide any additional
assistance.
Regards,
VoicePulse Customer Support
I'm running:
/usr/src/asterisk/asterisk -r
Asterisk CVS-HEAD-08/01/04-22:51:56, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
Connected to Asterisk CVS-HEAD-08/01/04-22:51:56 currently running on skip (pid = 2522)
skip*CLI
My /etc/asterisk/extensions.conf does:
exten = _1NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
[voicepulse-incoming]
exten=510740,1,Ringing
exten=510740,2,Wait,3
exten=510740,3,Answer
exten=510740,4,Agi,/usr/local/mipl/agnese|http://www..com/X.cgi?source=${EXTEN}callerid=${CALLERIDNUM}
exten=510740,5,Hangup
+++
Is there anyone else with a similar problem? A working setup?
-Bryce
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RE: [Asterisk-Users] need help with zaptel.
Ok I am past the compile. Now when I try to modprobe I get FATAL: Error inserting zaptel (/lib/modules/2.6.5-1.358/misc/zaptel.ko): Invalid module format -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Blakely Sent: Saturday, August 21, 2004 7:00 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] need help with zaptel. Look in /usr/src. You should see a directory something similar to linux-2.6.1-1[as an example]. If you DON'T have a directory (or link to a directory) named linux-2.6, you should create one using the 'ln' command. In the case mentioned above, the command would be: ln -s /usr/src/linux-2.6.1-1 /usr/src/linux-2.6 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edward Huitt Sent: Saturday, August 21, 2004 5:42 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] need help with zaptel. I cant get zaptel to make. I get this error: make: *** [linux26] Error 1. The previous line is: Link /usr/src/linux-2.6 to your kernel sources first! I am running Fedora Core2 Asterisk compiles fine. I am using my SIP phones. I would like to get my TDM400p working.
[Asterisk-Users] Uniden UIP200 Review
Brian, Im new to this list, and ran across this post about a Uniden UIP200. Since its been a few months now, I was wondering how it's turned out so far. I am also looking to implement * for a PBX, and am in search of a good hardware phone, that *hopefully* doesn't break the bank (HINT HINT :: All you other guys out there, feel free to chip in your opinions on what there is out there for good phones ! :-) ) Thanks, Tim + REPLIED TO + Hello Everyone, My company is about to deploy * as replacement for our existing commercial Altigen PBX. Meanwhile, I've been trying to find the best cost effective SIP VoIP phone which we can use in office for 20-30 employees, as well as a few remote staff. Normally I wouldn't post about a VoIP phone, however, this phone was released less than a week so I thought I'd give some feedback from an office perspective on the new unit. It is Uniden's first offering into the VoIP market. Main Features which were important to me: Built in 10/100 Switch Speakerphone w/headset port IEEE 802.3af Standard Inline Power (PoE) 2 line 16/char LCD Display 8 Programmable (not soft) Keys QoS [IEEE 802.1 p/q Based and DiffServ G711a/u G729A Codec Support TFTP Auto Configuration Firmware Upgrades (based on mac addressed filenames) The phone also has all the hard buttons you'd expect it to have. Hold, speaker/headset, Volume up and down, Menu, Transfer, Cancel, and Dial (used in lieu of pressing the # key to cut down digit timeouts when on-hook dialing). First, this phone, is relatively inexpensive. I was able to pick one up for $129. Setup and configuration was trying, as the phone ships with absolutely NOTHING in terms of an admin guide. The support areas on the Uniden site were password protected and even the support staff was unaware of all the proper logins and passwords (gotta love supporting new products). Once I gained access to the appropriate admin guide, I whipped up a few of the configuration files on my TFTP server, plugged in the phone and was off and rolling. Or so I thought. There seems to be some minor DHCP issues with the phone currently. It was ignoring my DHCP server's DHCP Offer's and constantly reported DHCP Failed on the LCD. After speaking with a Uniden Developer and sending him an ethereal trace, I hard-coded the IP address to continue my testing. The phone fired up, auto-configured itself via TFTP, and was logged into * in a matter of seconds. Needless to say, at this point, I was extremely pleased to see it actually WORKED. Weak Points: Wimpy Speakerphone: It's extremely easy for the speakerphone itself to over modulate. The microphone however does seem to perform well, even if it is a *little* tin-can'ish. Hold Button: Works as expected, * puts the caller on hold, and they hear MOH. YOU on the other hand hear this really cheesy Nintendo style genre of music locally, produced by the phone. When using speakerphone and placing someone on hold, this is extremely irritating. DTMF: When you have a session, or call active, there is no local DTMF feedback over the handset or speakerphone. While I'm ok with this, I can just picture my entire office on the first day, wondering if they actually pushed the buttons hard enough. So navigating through auto attendant menus can be a little tricky since you're not sure if you actually missed the button, or made solid contact. You can however check the LCD to see if the number you pressed went through. Conclusion: In testing, the phone is an all around solid performer. If they resolve my DHCP issue, I think we probably will go ahead and purchase 20-30 phones to start so that we can get * deployed in the near future. For $130, I don't think I can really complain about the weak points, however I have voiced my opinion on the DTMF and HOLD music to Uniden, so maybe in the near future we'll have some toggles in the TFTP config files make life a little less stressful. Uniden currently has a distributor/wholesaler who will sell to the public. If you're interested in picking up any of these phones to test yourself, the contact information is below. Note: Please keep in mind, Uniden also makes the UIP300 and UIP312. These phones *only* support H323. The UIP400 is the equivalent model of the 300, but will support SIP and is currently in development. Contact: Aimee @ Teledynamics (800) 847-5629 ext.110 or, [EMAIL PROTECTED] Brian D'Arcy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zultys 4x4 or 4x5
Michael Graves wrote: A friend has asked me to get a new Zultys 4x5 working with his * server. I've been over the web gui but don't see where all the registration info gets entered in the phone. Is there someone on-list who has Zultys phones and can advise me as to their setup? The manufacturers docs are useless. I've only used the 4x4, the username is the MAC address and the password is set using the phone keypad, a much more painful process then the bt101's... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers I do not try to dance better than anyone else. I only try to dance better than myself. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
