RE: [Asterisk-Users] Transfering from a device to a queue crashesAsterisk
Hi David, I've got probably the same/a similarproblem. Do you add the phones to the queue (AgentLogin/AddQueueMember)? If there are entries like: " Spawn extension (macro-dialout-trunk,s,21) exited non-zero..." in your * log file you might have the same problem like me. I suspect that something goes wrong with the nested macro calls within the AMP-generated dialplan, so what I did was to expand macro-dial etc.. for each local SIP extension. This seems to work, but is not really nice because you will have to configure the extensions manually (which makes AMP more or less obsolete)... cheers Jörg From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David F. BakkerSent: Thursday, September 15, 2005 5:28 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Transfering from a device to a queue crashesAsterisk Asterisk crashes with no errors when I transfer from a device (my phone)to a queue Asterisk crashes with no errors. Also if I xfer from a sip device to another and dont wait for the other user to pickup before xfering the call gets dropped. Any ideas? Im using the latest cvs of asterisk, amp 1.10.009 and our phones are polycom 501. Yahoo! for GoodClick here to donate to the Hurricane Katrina relief effort. ___ --Bandwidth and Colocation sponsored by Easynews.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] RxFax/TxFax - Compile Problem
On Thu, 2005-09-15 at 16:10 -0400, David Sampson wrote: I’ve reduced my problem down to this: [EMAIL PROTECTED]:/usr/src/asterisk/asterisk-1.0.9/apps# make Are you trying to use make from the apps directory? You have to run make from the main asterisk source directory. Look at the patch file necessary for the main Makefile. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.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] Asterisk and Zyxel Prestige 2000W_v2
Here is conf example [51] type=friend username=Test secret=testpassword host=dynamic canreinvite=no context=sip disallow=all allow=alaw dtmfmode=rfc2833 And you have to make phone conf. Like this Username = 51 Password = testpassword Phone number 51 [EMAIL PROTECTED] Iyi Calismalar. Ugur GUNCER System Administrator TeleBizz Telecommunication, Billing, Internet Satellite Solutions Akinci Bayiri Sokak No:13 Cevre Apt. B Blok Kat:3 Daire:5 Mecidiyekoy-Sisli/ISTANBUL Mobile:+90 544 535 97 37 Tel :+90 212 347 69 59 Fax :+90 212 347 69 49 Personal E-mail: [EMAIL PROTECTED] General E-mail: [EMAIL PROTECTED] Internet URL: http://www.telebizz.com.tr The information contained in this message is confidential and is intended for the addressee only. Any unauthorized dissemination or copying or use or disclosure of information contained herein is strictly prohibited and may be illegal. If you are not the named or intended recipient please notify us immediately by telephone, fax or return e-mail. TeleBizz has installed active virus software but does not accept liability or responsibility for the security or reliability of transmission or for any virus transmitted. The contents of this e-mail and any attachments are not intended as and do not constitute advice and TeleBizz disclaims any liability or responsibility for the accuracy thereof. Bu e-posta ve onunla iletilen bütün dosyalar sadece göndericisi tarafından alması amaçlanan yetkili gerçek ya da tüzel kişinin kullanımı içindir. Eğer söz konusu yetkili alıcı değilseniz bu elektronik postanın içeriğini açıklamanız, kopyalamanız, yönlendirmeniz ve kullanmanız kesinlikle yasaktır ve bu elektronik postayı derhal silmeniz gerekmektedir. TeleBizz bu mesajın içerdiği bilgilerin doğruluğu veya eksiksiz olduğu konusunda herhangi bir garanti vermemektedir. Bu nedenle bu bilgilerin ne şekilde olursa olsun içeriğinden, iletilmesinden, alınmasından ve saklanmasından sorumlu değildir. Bu mesajdaki görüşler yalnızca gönderen kişiye aittir ve TeleBizz'in görüşlerini yansıtmayabilir. Bu e-posta bilinen bütün bilgisayar virüslerine karşı taranmıştır. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus Sonnenleiter Sent: Friday, September 16, 2005 5:01 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk and Zyxel Prestige 2000W_v2 Hi Has anybody tried to use the Zyxel Prestige 2000W_v2 with Asterisk? I have the latest firmware and can now make outbound calls. So it looks like Asterisk does accept the configuration in sip.conf. However, I cannot receive any calls. Also, I keep getting this message on the console: chan_sip.c:7733 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '10.99.1.151'. TIA Klaus ___ --Bandwidth and Colocation sponsored by Easynews.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 BEGIN:VCARD VERSION:2.1 N:GUNCER;Ugur FN:Ugur GUNCER ([EMAIL PROTECTED]) ORG:Telebizz Telekomunikasyon ve Internet Hizmetleri Ltd. Sti.;IT TITLE:Sistem Yoneticisi TEL;WORK;VOICE:+90 (212) 347 69 59 TEL;CELL;VOICE:+90 (544) 535 97 37 TEL;WORK;FAX:+90 (212) 347 69 49 ADR;WORK;ENCODING=QUOTED-PRINTABLE:;Telebizz Istanbul;Akinci Bayiri=0D=0ASokak No:13 Cevre Apt. =0D=0AB Blok Da= ire: 5;Mecidiyekoy-Sisli, Istanbul;;80290;T=FCrkiye LABEL;WORK;ENCODING=QUOTED-PRINTABLE:Telebizz Istanbul=0D=0AAkinci Bayiri=0D=0ASokak No:13 Cevre Apt. =0D=0AB Blo= k Daire: 5=0D=0AMecidiyekoy-Sisli, Istanbul 80290=0D=0AT=FCrkiye EMAIL;PREF;INTERNET:[EMAIL PROTECTED] REV:20050606T102426Z END:VCARD smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation sponsored by Easynews.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] auto restart
Hello, I have see a post in the list saying that the 'daemon' command should be remove from the asterisk startup script in /etc/rc.d/init.d/ for FC2 in order for asterisk to auto restart when crash. I wonder if this should be done on FC3 as well, because my asterisk did not restart when crash. Please help Thanks CCF ___ --Bandwidth and Colocation sponsored by Easynews.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] Wildcard TE110P
Hi; I have a Wildcard TE110P, i'm locking for a Motherboard to use it, i know that in the web site of Digium are 3 Motherboard but it's to expensive. Do you tested other Motherboard. Thanks -- coordialement Karim AMER ___ --Bandwidth and Colocation sponsored by Easynews.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] Unable to create ZAP channel - All circuits are busy
Hello,
I have [EMAIL PROTECTED] 1.5 installed and all is working fine for
incoming calls and sometimes outgoing calls. Installed in the box is a digium
TDM04B (4xFXO Ports)
setup as ZAP1 to
ZAP4. I have incoming calls coming in on lines 1-4 in that order and
outgoing calls prefering ZAP4 then ZAP3 then ZAP2.
When i try to dial
out to the PSTN from a SIP phone it sometimes works (normally after a reboot)
and a few calls later i get the voiceprompt "All circuits are
busy."
I know the PSTN line
is on hook and ok to use.
Looking in the log
file i can see a line saying unable to create a ZAP channel. (See extract
below)
Sep 16 07:47:46 VERBOSE[2012]: -- Executing
SetVar("SIP/200-361a", "DIAL_NUMBER=850220") in new stackSep 16 07:47:46
VERBOSE[2012]: -- Executing SetVar("SIP/200-361a", "DIAL_TRUNK=3") in new
stackSep 16 07:47:46 VERBOSE[2012]: -- Executing AGI("SIP/200-361a",
"fixlocalprefix") in new stackSep 16 07:47:46 VERBOSE[2012]: -- Launched AGI
Script /var/lib/asterisk/agi-bin/fixlocalprefixSep 16 07:47:46
VERBOSE[2012]: fixlocalprefix: Could not parse
/etc/asterisk/localprefixes.confSep 16 07:47:46 VERBOSE[2012]: -- AGI Script
fixlocalprefix completed, returning 0Sep 16 07:47:46 VERBOSE[2012]: --
Executing SetVar("SIP/200-361a", "OUTNUM=850220") in new stackSep 16
07:47:46 VERBOSE[2012]: -- Executing Cut("SIP/200-361a", "custom=OUT_3|:|1") in
new stackSep 16 07:47:46 DEBUG[2012]: _expression_ is '0'Sep 16 07:47:46
VERBOSE[2012]: -- Executing GotoIf("SIP/200-361a", "0?19") in new stackSep
16 07:47:46 DEBUG[2012]: Not taking any branchSep 16 07:47:46 VERBOSE[2012]:
-- Executing Dial("SIP/200-361a", "ZAP/2/850220") in new stackSep 16
07:47:46 NOTICE[2012]: Unable to create channel of type 'ZAP'Sep 16
07:47:46 VERBOSE[2012]: == Everyone is busy/congested at this timeSep 16
07:47:46 DEBUG[2012]: Exiting with DIALSTATUS=CHANUNAVAIL.Sep 16 07:47:46
VERBOSE[2012]: -- Executing Goto("SIP/200-361a", "s-CHANUNAVAIL|1") in new
stackSep 16 07:47:46 VERBOSE[2012]: -- Goto
(macro-dialout-trunk,s-CHANUNAVAIL,1)Sep 16 07:47:46 VERBOSE[2012]: --
Executing NoOp("SIP/200-361a", "Dial failed due to CHANUNAVAIL") in new
stackSep 16 07:47:46 VERBOSE[2012]: -- Executing Macro("SIP/200-361a",
"outisbusy") in new stackSep 16 07:47:46 VERBOSE[2012]: -- Executing
Playback("SIP/200-361a", "allison7/all-circuits-busy-now") in new stackSep
16 07:47:46 DEBUG[2012]: Ooh, format changed from unknown to ulawSep 16
07:47:46 DEBUG[2012]: Scheduling timer at 160 sample intervalsSep 16
07:47:46 VERBOSE[2012]: -- Playing 'allison7/all-circuits-busy-now' (language
'en')Sep 16 07:47:46 DEBUG[2012]: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 102: FoundSep
16 07:47:47 DEBUG[2012]: Setting NAT on RTP to 0Sep 16 07:47:48 DEBUG[2012]:
Scheduling timer at 0 sample intervalsSep 16 07:47:48 VERBOSE[2012]: ==
Spawn extension (macro-outisbusy, s, 1) exited non-zero on 'SIP/200-361a' in
macro 'outisbusy'Sep 16 07:47:48 VERBOSE[2012]: == Spawn extension
(from-internal, 9850220, 4) exited non-zero on 'SIP/200-361a'Sep 16 07:47:48
VERBOSE[2012]: -- Executing Macro("SIP/200-361a", "hangupcall") in new
stackSep 16 07:47:48 VERBOSE[2012]: -- Executing ResetCDR("SIP/200-361a",
"w") in new stackSep 16 07:47:48 DEBUG[2012]: cdr_mysql: inserting a CDR
record.
Ive tried rebooting, genzaptel,
rebuilt zaptel drivers etcDoes anyone have any suggestions as
to what i can do to debug.
Any help would be
appreciated.
Thanks
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Re: [Asterisk-Users] Call recording between SIP phones
canreinvite=no - Original Message - From: Kevin Bockman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, September 15, 2005 9:48 PM Subject: Re: [Asterisk-Users] Call recording between SIP phones Lakmal wrote: / I have been searching around for days on how to record calls between SIP/ / Phones. Could someone tell me whether it is possible? The Record command/ / doesn't seem to work during a call./ If you are using a fairly recent version of HEAD, you can check out Dial options w or W and features.conf. This will enable a feature code for you to press while you are on the call to start the recording. Kevin ___ --Bandwidth and Colocation sponsored by Easynews.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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.25/102 - Release Date: 9/14/05 ___ --Bandwidth and Colocation sponsored by Easynews.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] Call Forward - 7940 Asterisk - Help
I am looking for a simple way to forward calls unconditionally with Asterisk. We are running an Asterisk system with 10 extensions using SIP. One of our users leaves the office regulary,when she is out,she needs to be able to forward unconditionally to her mobile or collegue. I am trying to keep it as simple as possible, we use Cisco 7940's, they have a call forward option, when she uses it, all our incoming calls go to her mobile? Not just the calls to her extension. My Question: Does Call Forward on the Cisco Phones and Asterisk work? If so do I need to implement something into the dial plan. I have read on the wiki:http://www.voip-info.org/tiki-index.php?page=Asterisk%20call%20forwarding Is there an updated way to do this? I thought *21* was hard coded into Asterisk? If the Cisco phones wont work, i would like her to simply dial *21*mobile number#, any suggestions on this? Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.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] MusicOnHold not working
Hi, I am using asterisk 1.0.9 on FC3 box. mpg123 is not working it starts and stops immediately.. printing - -- Started music on hold, class 'default', on channel 'Zap/1-1' -- Stopped music on hold on Zap/1-1 Unknown option: --mono There is no such option defined in my musiconhold.conf file After searching through code: I got --mono option defined in function spawn_mp3(struct mohclass *class) in file res_musiconhold.c After commenting that line message stopped but mpg123 still didn't responded nd I am trying to find. I am wondering if mpg123 doesnot support --mono option They why it is forced in asterisk Code? Thanks Gurminder On 9/15/05, Alex Kobalto [EMAIL PROTECTED] wrote: I have the same problem with several softphones (Xlite), but there's one, Firefly I think, that worked. I found it strange, but not a real problem for me. I have the same asterisk server version, wich came with the last [EMAIL PROTECTED] distribution. On 9/15/05, Sherwood McGowan [EMAIL PROTECTED] wrote: It's because mpg123 is being passed an option --mono. Looks to me (a cursory guess) that your asterisk installation is trying to force mono sound, and mpg123 doesn't like it. --Original Message- -From: [EMAIL PROTECTED] -[mailto: [EMAIL PROTECTED] On Behalf Of -Gurminder Arora -Sent: Thursday, September 15, 2005 8:30 AM -To: asterisk-users@lists.digium.com -Subject: [Asterisk-Users] MusicOnHold not working - -Hi - On my FC3 box with asterisk 1.0.9MusicOnHold is not working. -It starts and stops immediately... -An unknow option mono comes...from where it is originating.?? -As there is nothing written in .conf file. -Console output is below: - - -I am using mpg123 version 0.59r. -Although I am able to play music with mpg123 but why it is on -No-cooperation movement against asterisk ? - -Need help..any ideas any direction... - -Thanks -Gurminder - - -Console output -***Snip*** --- Executing MusicOnHold(Zap/1-1, default) in new stack --- Started music on hold, class 'default', on channel 'Zap/1-1' --- Stopped music on hold on Zap/1-1 -Unknown option: --mono -*** - - - -My musiconhold.conf is -***Snip -[default] -mode=mp3 -directory=/var/lib/asterisk/mohmp3 -application=/usr/local/bin/mpg123 -Snip** -___ ---Bandwidth and Colocation sponsored by Easynews.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 - ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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
use of automon sequence (was Re: [Asterisk-Users] Call recording between SIP phones)
OK, I wonder if I have something wrong -- I have the *1 in my features.conf for the one touch record -- now I called a number, and when the call was answered flashed the hook and pressed *1 and went back tothe call, but nothing happpened. I am using CVS from 8/26 -- is this too old or am I doing something wrong? I did add wW to the dial options of the call and that part did work, but there is nothing in the log about automon. Also, what kind of file does this produce, etc? Thanks in advance for any help. on Thursday 09/15/2005 Kevin Bockman([EMAIL PROTECTED]) wrote Lakmal wrote: / I have been searching around for days on how to record calls between SIP/ / Phones. Could someone tell me whether it is possible? The Record command/ / doesn't seem to work during a call./ If you are using a fairly recent version of HEAD, you can check out Dial options w or W and features.conf. This will enable a feature code for you to press while you are on the call to start the recording. Kevin ___ --Bandwidth and Colocation sponsored by Easynews.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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.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] How to suppress Local/Zombie channels?
hi, can anyone please tell me under which circumstances asterisk creates Local/Zombie channels and how to suppress this? It only seems to happen when a user calls himself, but I can't reproduce this in our testsystem and it only happens occasionally. All we do in the extensions.conf is send the incoming sip call control out to a FastAGI server, which does the actual Dial command. We don't consciously create any Local channels, its definitely asterisk who decides that theres a need for it ... Any help appreciated, thanks a lot, Tony ___ --Bandwidth and Colocation sponsored by Easynews.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] timeout with queue SOLVED
In extensions.conf I have changed the queue command from:
exten = 1,1,Queue(itsupport|n|||50)
to:
exten = 1,1,Queue(itsupport|tT|||50)
Now it works as it should.
Regards
Wolfgang
Am Donnerstag, 15. September 2005 08:21 schrieb Wolfgang Lumpp:
Am Mittwoch, 14. September 2005 19:12 schrieb Sander:
In queues.conf
; How long do we let the phone ring before we consider this a timeout...
;
timeout = 15
This is set in queues.conf
But this is just the function how long the phones will ring you should
not set this option to long or your phone will stop ringing if a timeout
is set in your phone
But when the line hangs up after timeout you have set an option at the
queue like this below it will stay in queue for 15 seconds then hangs up
exten = 121,2,Queue(121|tT|||15)
Here I have tried with 30 up to 120
Every time after the second Nobody picked up in 15000ms I get the
Exiting on time-out cycle Stopped music on hold on CAPI/contr1/s-0
So the 3rd Queue member isn't called.
Below are the configs
Hi,
I've setup a queue with 3 sip members.
I've tried with random and roundrobin and different timeout settings
in musiconhold.conf Always after the second Nobody picked up in 15000ms
I get Exiting on time-out cycle Stopped music on hold on
CAPI/contr1/s-0
Where can I increase this timeout?
asterisk 1.0.9 on linux 2.6.11 SuSE 9.3
queues.conf
[general]
[default]
[itsupport]
music = default
strategy = roundrobin
retry = 5
maxlen = 5
member = sip/980
member = sip/981
member = sip/982
extensions.conf
[general]
static=yes
writeprotect=yes
[globals]
OUTBOUND=CAPI/contr1/
WOL=SIP/980
JSK=SIP/981
SGA=SIP/982
EVERYONE=${WOL}${SGA}${JSK}
[default]
include = mainmenu
[outbound]
include = externalcalls
include = international
[inbound-from-local]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
include = internalextensions
include = outbound
[inbound-from-sip]
include = default
[international]
exten = _00.,1,Congestion
exten = _14100.,1,Congestion
[externalcalls]
exten = _0.,1,Dial(${OUTBOUND}/b0${EXTEN},40,tT)
exten = _0.,2,Congestion
exten = _XXX,1,Dial(${OUTBOUND}/b${EXTEN},40,tT)
exten = _XXX.,2,Congestion
[inbound-from-sip]
include = default
[internalextensions]
exten = 980,1,Macro(stdexten,980,${WOL})
exten = wol,1,Goto(980|1)
exten = 981,1,Macro(stdexten,981,${JSK})
exten = jsk,1,Goto(981|1)
exten = 982,1,Macro(stdexten,982,${SGA})
exten = sga,1,Goto(982|1)
[macro-stdexten]
exten = s,1,Playback(transfer)
exten = s,2,Dial(${ARG2},25,m)
exten = s,3,Hangup()
[mainmenu]
exten = s,1,Wait,1 ; Wait a second, just for fun
exten = s,2,Answer ; Answer the line
exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = s,4,ResponseTimeout,130 ; Set Response Timeout to 10
seconds
exten = s,5,BackGround(fallsenglisch) ;
exten = s,6,BackGround(willkommen) ;
exten = s,7,BackGround(probleme) ;
exten = s,8,BackGround(it-menu)
exten = s,9,Goto(s,5)
exten = 1,1,Queue(itsupport|n|||50)
exten = 2,1,Goto(s,5)
exten = 5,1,BackGround(willkommen-en)
exten = 5,2,BackGround(probleme-en)
exten = 5,3,BackGround(it-menu-en)
exten = 5,4,Goto(5,1)
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Re: [Asterisk-Users] USB Phones for use with Asterisk
Are there drivers available for Xten Softphones? Do they work with X-lite/Eyebeam Softphones? On 9/16/05, Bill McCready (PCPhoneline.com) [EMAIL PROTECTED] wrote: Hi all, I have a question that I was hoping someone could answer for me. I would like to find a USB phone that works with Asterisk... preferably one that does not have any issues with NAT.RESPONSE:This is Bill McCready from PCPhoneline.com . I suggest you giveour VTA1000 Skype+SIP+IAX2 Gateway which is USB based with a good cordlessphone. We offer several versions of the VTA1000 but this is the latest one we havedeveloped and it supports Skype + 10 x SIP + 10 x IAX2 networks all at thesame time and plugs into the USB port of a Windows 2000 or Windows XP computer. You can optionally disable Skype and/or the SIP callingcapabilities if you don't need them. The VTA1000 is designed to work evenif your computer screen is powered off using it with a cordless phone of course gives you the added advantage of not needing to be tied to yourcomputer to place and receive calls. If you don't mind being tethered to acomputer we also have the VPT1000 phone available with the same capability set.You can find them in our Ebay store at http://stores.ebay.com/pcphoneline .If you need a bunch of these and want a build specially tailored and branded please private e-mail me with your requirements and we will do our best totry to accomodate you.Best regards...Bill Can anyone point me to something suitable ?We are essentially planning to do a beta for a few hundred customers on our network and would like them to be able to call one another @ no charge. Because these are going to be end users who have no idea about Voice / VOIP stuff... it needs to be extremely easy to use. Can someone point me in the right direction ? Cheers, Callum ___ --Bandwidth and Colocation sponsored by Easynews.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___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.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] Call Forward - 7940 Asterisk - Help
c waddy schrieb: I am looking for a simple way to forward calls unconditionally with Asterisk. We are running an Asterisk system with 10 extensions using SIP. One of our users leaves the office regulary, when she is out, she needs to be able to forward unconditionally to her mobile or collegue. I am trying to keep it as simple as possible, we use Cisco 7940's, they have a call forward option, when she uses it, all our incoming calls go to her mobile? Not just the calls to her extension. My Question: Does Call Forward on the Cisco Phones and Asterisk work? If so do I need to implement something into the dial plan. I have read on the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20call%20forwarding Is there an updated way to do this? I thought *21* was hard coded into Asterisk? If the Cisco phones wont work, i would like her to simply dial *21*mobile number#, any suggestions on this? AFAIK, the CFWDALL option of the SIP fw send a temporarily moved message back to the caller with the new address to call. This can fail for a lot of reasons. I would recommend using 7940s with the native Skinny firmware and chan_sccp by Sergio Chersovani. Chan_sccp not only supports call forwarding but a lot more and only with the Skinny image, you can use all the features, the phone have. You can read a bit more at http://chan-sccp.org/ Regards, Stefan -- (o_ Stefan Gofferje| SCLT, MCP //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ --Bandwidth and Colocation sponsored by Easynews.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] Call Forward - 7940 Asterisk - Help
Hi Stefan, Has chan_sccp matured a lot? I remember (maybe a year ago or so) that I had a lot of problems with chan_sccp and chan_skinny (one thing is that I remember with chan_sccp is that the VM button didn't work and trying to answer multiple incoming calls tended to make the phone go into a weird state where I had to power cycle it to get it back right again). I upgraded the phoen to SIP and never looked back - but then maybe I'm not getting all the fetures like call forwarding (which sounds very useful to me).. Derek Stefan Gofferje wrote: I would recommend using 7940s with the native Skinny firmware and chan_sccp by Sergio Chersovani. Chan_sccp not only supports call forwarding but a lot more and only with the Skinny image, you can use all the features, the phone have. You can read a bit more at http://chan-sccp.org/ -- Derek Conniffe Rivertower Ltd Ireland: (Freephone) 1800 719 400 Ireland: (Local) 01 244 9719 United Kingdom: 0870 068 2368 International: 00 353 1 244 9719 Derek Conniffe DDI: 01 201 0146 (International: 00 353 1 201 0146) Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 856 3823) Fax: 01 201 0085 (International: 00 353 1 201 0085) Email: [EMAIL PROTECTED] Web: http://www.rivertowerhosting.com begin:vcard fn:Derek Conniffe n:Conniffe;Derek org:Rivertower Ltd;IT adr:Dublin 2;;46 Upper Mount Street;Dublin;Dublin;Dublin 2;Ireland email;internet:[EMAIL PROTECTED] tel;work:+353 1 201 0146 tel;fax:+353 1 201 0085 tel;cell:+353 86 856 3823 note;quoted-printable:Ireland: (Freephone) 1800 719 400=0D=0A= Ireland: (Local) 01 244 9719=0D=0A= United Kingdom: 0870 068 2368=0D=0A= International: 00 353 1 244 9719=0D=0A= url:http://www.rivertowerhosting.com version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.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] 7 digit dialing to e.164 format
All, I've asked this once a long time ago and got a vague response, any
suggestions? I'm wanting to convert for example a 7 digit extension
(whether it be via dialplan or agi) to e.164. This is for the sake of
getting everything outbound into e164 format. The issue I see you will
need to append the areacode of the calling party to the 7 digits, from
there adding a +1 is of course easy. example:
Customer dials 5610680, his callerid (from) is +16364424591.. I want to
take the 636 and add it to the 561 number so the result would be
6365610680 .. Any thoughts? I've been looking at the internal ast
commands with no luck. I'm sure people have run into this before, it
would be nice to add a ${e164_fromNPA} variable!!! ;-)
Matt
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Re: [Asterisk-Users] Is digium supporting new te405p and te406p install?
On Thursday 15 September 2005 23:54, Jason Kim wrote: I tried both 1.0.9 and 1.2beta. I couldn't see any interrupt from /proc/interrupt. My email server has no spam filter. rmmod wct4xxp zaptel (ignore any errors) dmesg -c /dev/null modprobe wct4xxp dmesg -c /tmp/dmesg.wct4xxp what is the contents of /tmp/dmesg.wct4xxp? -A. ___ --Bandwidth and Colocation sponsored by Easynews.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] Unable to call some numbers with I4L
Thank you for your response. Can i make the same configuration with I4L (how?) ? The EICON DIVA PCI 2.02 (as i know) is not usable with CAPI, but only with I4L. Am i wrong ? Massimo Frisoni Emanuele Pucciarelli wrote: Massimo Frisoni ha scritto: I have an EICON DIVA PCI 2.02 with I4L. I'm unable to call some numbers, in general numbers with automatic responders that do not rings. It's seems asterisk does not understand that the other party has answered, so after a timeout it reports 'busy', but in real the other end has answered. Any other call to a normal number works fine. You should probably try chan_capi and enable inband call progress reporting. (IIRC, that's done by putting b in the dialstring before the dialed number!) I wouldn't know how the same thing is done through I4L, maybe somebody else can shed light on that... Bye, ___ --Bandwidth and Colocation sponsored by Easynews.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] Help on RealTime Extensions on Oracle DB
hello
I'm working with realtime and oracle.
I'm using two tables in oracle (sip_conf and voicemail_conf)
My extensions.conf looks like this:
[datab]
exten = _3XXX,1,Dial(SIP/${EXTEN})
exten = _3XXX,2,Voicemail(u${EXTEN})
exten = _3XXX,3,Hangup
exten = _3XXX,104,Voicemail(b${EXTEN})
exten = _3XXX,105,Hangup
switch = Realtime
It's work very well!
Regards.
JS.
-Mensaje original-
De: Chris Deserva [mailto:[EMAIL PROTECTED]
Enviado el: Friday, September 16, 2005 1:27 AM
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] Help on RealTime Extensions on Oracle DB
Does someone here configured RealTime Extensions using
ODBC connecting to Oracle DB? Im having a problem in
dialplan patterns, it doesnt work. Pls. help!
-Chris
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[Asterisk-Users] Asterisk don't start
Asterisk don't running, because show this message Sep 16 15:28:42 WARNING[1075709024]: cli.c:702 ast_cli_register: Command 'showparkedcalls' already registered (or something close enough) == Parsing '/etc/asterisk/features.conf': Found -- Added extension '700' priority 1 to parkedcalls /usr/sbin/asterisk: relocation error: /usr/lib/asterisk/modules/res_features.so: undefined symbol: ast_pthread_create THANKS! ___ --Bandwidth and Colocation sponsored by Easynews.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] RxFax/TxFax - Compile Problem
If I understand correctly you are supposed to patch the Makefile in the apps directory and then run the main Makefile. I've tried both ways - the patch failed on the main Makefile. Should I try to make that work? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Friday, September 16, 2005 3:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RxFax/TxFax - Compile Problem On Thu, 2005-09-15 at 16:10 -0400, David Sampson wrote: I’ve reduced my problem down to this: [EMAIL PROTECTED]:/usr/src/asterisk/asterisk-1.0.9/apps# make Are you trying to use make from the apps directory? You have to run make from the main asterisk source directory. Look at the patch file necessary for the main Makefile. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Unable to call some numbers with I4L
On Fri, 16 Sep 2005, Massimo Frisoni wrote: Thank you for your response. Can i make the same configuration with I4L (how?) ? The EICON DIVA PCI 2.02 (as i know) is not usable with CAPI, but only with I4L. Am i wrong ? I don't know the current status, but mISDN and its CAPI should support it. Armin Emanuele Pucciarelli wrote: Massimo Frisoni ha scritto: I have an EICON DIVA PCI 2.02 with I4L. I'm unable to call some numbers, in general numbers with automatic responders that do not rings. It's seems asterisk does not understand that the other party has answered, so after a timeout it reports 'busy', but in real the other end has answered. Any other call to a normal number works fine. You should probably try chan_capi and enable inband call progress reporting. (IIRC, that's done by putting b in the dialstring before the dialed number!) I wouldn't know how the same thing is done through I4L, maybe somebody else can shed light on that... Bye, ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Call Forward - 7940 Asterisk - Help
Hi Derek Derek Conniffe schrieb: Hi Stefan, Has chan_sccp matured a lot? I remember (maybe a year ago or so) that I had a lot of problems with chan_sccp and chan_skinny (one thing is that I remember with chan_sccp is that the VM button didn't work and trying to answer multiple incoming calls tended to make the phone go into a weird state where I had to power cycle it to get it back right again). I upgraded the phoen to SIP and never looked back - but then maybe I'm not getting all the fetures like call forwarding (which sounds very useful to me).. I am talking about a complete rewrite. About half a year ago, Sergio was writing patches for Julien's chan_sccp (chan-sccp.sf.net). For some reason, Sergio decided to do a project split and started his own chan_sccp. After a few weeks he stopped patching Julien's work and did a complete rewrite. Meanwhile, chan_sccp (chan-sccp.berlios.de or chan-sccp.org) is not only stable but also very feature reach! Sergio does the coding and me and some other people do very heavy testing and the project is making great progress. Highlights are: - Line status monitoring on 7960/7914 (you see what other - not only SCCP - extensions are doing) - Good hardware support (7940/7960/7914 almost 100%, 7905 almost 90%, 7920 almost 80%, 7970 currently under heavy development) - Support for call waiting, call forwarding, ... - Intercom / autoanswer - controlled by dial application and a lot more... Have a look at http://chan-sccp.org/ ... Slan go foil, Stefan -- (o_ Stefan Gofferje| SCLT, MCP //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface ___ --Bandwidth and Colocation sponsored by Easynews.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] Call Forward - 7940 Asterisk - Help
Hi I am only interested in using SIP, i would like to setup simple call forwarding either by thephone or Asterisk, it is a commonlegacy PBX feature and i am sure it is available from Asterisk. What would be the best way to do it? Is it hard coded into Asterisk? Why do the SIP Cisco 7940's call forward feature not work with Asterisk? On 9/16/05, Stefan Gofferje [EMAIL PROTECTED] wrote: Hi DerekDerek Conniffe schrieb: Hi Stefan, Has chan_sccp matured a lot?I remember (maybe a year ago or so) that I had a lot of problems with chan_sccp and chan_skinny (one thing is that I remember with chan_sccp is that the VM button didn't work and trying to answer multiple incoming calls tended to make the phone go into a weird state where I had to power cycle it to get it back right again). I upgraded the phoen to SIP and never looked back - but then maybe I'm not getting all the fetures like call forwarding (which sounds very useful to me)..I am talking about a complete rewrite. About half a year ago, Sergio waswriting patches for Julien's chan_sccp (chan-sccp.sf.net).For some reason, Sergio decided to do a project split and started his own chan_sccp. After a few weeks he stopped patching Julien's work anddid a complete rewrite. Meanwhile, chan_sccp (chan-sccp.berlios.de or chan-sccp.org) is not only stable but also very feature reach!Sergio does the coding and me and some other people do very heavytesting and the project is making great progress.Highlights are:- Line status monitoring on 7960/7914 (you see what other - not only SCCP - extensions are doing)- Good hardware support (7940/7960/7914 almost 100%, 7905 almost90%, 7920 almost 80%, 7970 currently under heavy development)- Support for call waiting, call forwarding, ... - Intercom / autoanswer - controlled by dial applicationand a lot more...Have a look at http://chan-sccp.org/ ...Slan go foil,Stefan--(o_ Stefan Gofferje| SCLT, MCP //\ Reg'd Linux User #247167 | VCP #2263V_/_Heckler Koch - the original point and click interface___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.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] broadvoice incoming caller ID is wierd when calling from voipjet
Calling bv from pstn phones my log shows correct caller id number and name Calling from voipjet with cid set to 10 digit number(207826) my logs show bv adding a leading + and setting the name to egypt as a result Everything else I call from voipjet gets caller id correct Could be they are using the same carrier for caller ID to be mishandled this way. Anyone else noticed this behavior? ___ --Bandwidth and Colocation sponsored by Easynews.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] Extension Restrictions
Is it possible to define an extension that is not allowed to make or receive calls, unless an agent logs in? Obviously it would require that the extension be able to dial the # for the agent to log in. Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.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] Seperate Incoming calls on TDM02?
Yeah, in your zapata.conf just give each channel a different context setting. It's slightly more complicated with [EMAIL PROTECTED] and/or AMP, you need to use the zapata_custom.conf file, instead. You also need to use the extensions_custom.conf file, too, though there might be a better way I don't know about. I'm fighting with this right now and I'm hitting a serious frustration point - right now all incoming calls are getting handled by the from_pstn context which is how it honestly should be according to the current conf files. However when I change the context from from_pstn to aa_1 and aa_2 respectively it doesn't change anything in the way the system is answering the lines. Hatton zapata.conf: ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=no threewaycalling=no transfer=no cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include AMP configs #include zapata_additional.conf ;Include genzaptelconf configs #include zapata-auto.conf zapata_additional.conf is empty zapata-auto.conf: ; Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit ; Zaptel Channels Configurations (zapata.conf) ; ; This is not intended to be a complete zapata.conf. Rather, it is intended ; to be #include-d by /etc/zapata.conf that will include the global settings ; callerid=asreceived ; Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1 ; channel 1, WCTDM, inactive. ; channel 2, WCTDM, inactive. signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 3 context=from_pstn group=0 channel = 3 signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4 context=from_pstn group=0 channel = 4 ___ --Bandwidth and Colocation sponsored by Easynews.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] Asterisk don't start
Asterisk don't running, because show this message WARNING[1075709024]: cli.c:702 ast_cli_register: Command 'showparkedcalls' already registered (or something close enough) == Parsing '/etc/asterisk/features.conf': Found -- Added extension '700' priority 1 to parkedcalls /usr/sbin/asterisk: relocation error: /usr/lib/asterisk/modules/res_features.so: undefined symbol: ast_pthread_create Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.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] Asterisk don't start
Asterisk don't running, because show this message WARNING[1075709024]: cli.c:702 ast_cli_register: Command 'showparkedcalls' already registered (or something close enough) == Parsing '/etc/asterisk/features.conf': Found -- Added extension '700' priority 1 to parkedcalls /usr/sbin/asterisk: relocation error: /usr/lib/asterisk/modules/res_features.so: undefined symbol: ast_pthread_create Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.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] Caller Name: Asterisk reading too fast
I asked my telco to release caller name on the PRI. Earlier they were releasing only the phone number. I still did not see the name, but only the number in caller id. Actually I now see number twice. When I inquired with them this is the response I got: I ran a trace on your TG. I see that your switch is picking up the call so fast that it is not able to pick up the name. The name is being sent, but I suspect after it is too late. This is something that will need to be corrected in your switch. I have attached a sample call out of the trace I performed this morning. They have sent me the trace file. Is there a way as it is in Asterisk so that it reads the caller name properly? Thanks, -- jt ___ --Bandwidth and Colocation sponsored by Easynews.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] USB Phones for use with Asterisk
USB phone and NAT - What has USB Phpne got to do with NAT? USB Phone is just a hardware piece that pipes the audio output from your softphone. Your softphone has to take care of that. Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Callum McGillivray Sent: Thursday, September 15, 2005 8:41 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] USB Phones for use with Asterisk Hi all, I have a question that I was hoping someone could answer for me. I would like to find a USB phone that works with Asterisk... preferably one that does not have any issues with NAT. Can anyone point me to something suitable ? We are essentially planning to do a beta for a few hundred customers on our network and would like them to be able to call one another @ no charge. Because these are going to be end users who have no idea about Voice / VOIP stuff... it needs to be extremely easy to use. Can someone point me in the right direction ? Cheers, Callum ___ --Bandwidth and Colocation sponsored by Easynews.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 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.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] Caller Name: Asterisk reading too fast
Have you tried placing a Wait(1) before Answer() in your dialplan? On Friday 16 September 2005 11:23, J Thomas wrote: I asked my telco to release caller name on the PRI. Earlier they were releasing only the phone number. I still did not see the name, but only the number in caller id. Actually I now see number twice. When I inquired with them this is the response I got: I ran a trace on your TG. I see that your switch is picking up the call so fast that it is not able to pick up the name. The name is being sent, but I suspect after it is too late. This is something that will need to be corrected in your switch. I have attached a sample call out of the trace I performed this morning. They have sent me the trace file. Is there a way as it is in Asterisk so that it reads the caller name properly? Thanks, -- jt ___ --Bandwidth and Colocation sponsored by Easynews.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 -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation sponsored by Easynews.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] SIP port assignment for user agents registering to Asterisk.
I was wondering if anyone knows why when I register a user agent like XLite with Asterisk I am noticing that the port assignment on the sip show peers command shows the port to be different than any of the other user agents. The other user agents are logging in from different networks from all over the internet. Here is a sample of my table when I issue the command: Ipbx1*CLIsip show peers Name/username Host Dyn Nat ACL Mask Port Status cydalgw1/216.90.228.254 216.90.228.254 255.255.255.255 5060 OK (14 ms) steve/steve 68.113.199.172 D N 255.255.255.255 62784 Unmonitored hooman7/hooman7 68.88.66.248 D N 255.255.255.255 5060 UNREACHABLE I am using the xlite soft phone. I tried other phones but got the same result. Can anyone give me a clue as to what is going on? I figure my laptop might be screwed up or something. Steve ___ --Bandwidth and Colocation sponsored by Easynews.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] 7 digit dialing to e.164 format
You could name your peers by the full e.164 number associated with them
then parse the area code from that to append to the number they dialed
as a 7 digit number possibly.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Schulte
Sent: Friday, September 16, 2005 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] 7 digit dialing to e.164 format
All, I've asked this once a long time ago and got a vague response, any
suggestions? I'm wanting to convert for example a 7 digit extension
(whether it be via dialplan or agi) to e.164. This is for the sake of
getting everything outbound into e164 format. The issue I see you will
need to append the areacode of the calling party to the 7 digits, from
there adding a +1 is of course easy. example:
Customer dials 5610680, his callerid (from) is +16364424591.. I want to
take the 636 and add it to the 561 number so the result would be
6365610680 .. Any thoughts? I've been looking at the internal ast
commands with no luck. I'm sure people have run into this before, it
would be nice to add a ${e164_fromNPA} variable!!! ;-)
Matt
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Re: use of automon sequence (was Re: [Asterisk-Users] Call recording between SIP phones)
John covici wrote: OK, I wonder if I have something wrong -- I have the *1 in my features.conf for the one touch record -- now I called a number, and when the call was answered flashed the hook and pressed *1 and went back tothe call, but nothing happpened. I am using CVS from 8/26 -- is this too old or am I doing something wrong? I did add wW to the dial options of the call and that part did work, but there is nothing in the log about automon. Also, what kind of file does this produce, etc? Thanks in advance for any help. I'm not sure when it started, but as long as you have it in the example features.conf in your asterisk source/configs/features.conf file. You don't do a flash to do the record. You press the sequence WHILE YOU ARE ON THE CALL. It will record a file to /var/spool/monitor. To change the filename, you probably have to set the MONITOR_FILENAME variable before the dial. Kevin ___ --Bandwidth and Colocation sponsored by Easynews.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] Seperate Incoming calls on TDM02?
I'm fighting with this right now and I'm hitting a serious frustration point - right now all incoming calls are getting handled by the from_pstn context which is how it honestly should be according to the current conf files. However when I change the context from from_pstn to aa_1 and aa_2 respectively it doesn't change anything in the way the system is answering the lines. Found the source of the problem and it wasn't in the config files, rather in the way that Asterisk reloads when issued that command - it does NOT rebuild the Zapata channel table. I had to perform a complete restart of Asterisk to get the changes (which I made in my zapata-auto.conf file) into the system. After I did that everything started running right as rain! Thanks for the assistance, hopefully this will make it where people will see it in the future. Hatton ___ --Bandwidth and Colocation sponsored by Easynews.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] RxFax/TxFax - Compile Problem
I had sone trouble with this also.
But I got it.
I used the latest SpanDSP
0.0.9pre1
I used the applications app)_txfax and app_rxfax from the previous release
0.0.2pre8
I have included my makefile and the two app fiels that complied on the Latest
CVS (last night) on FC3.
Apps/MakeFile
#
# Asterisk -- A telephony toolkit for Linux.
#
# Makefile for PBX applications
#
# Copyright (C) 1999-2005, Digium
#
# Mark Spencer [EMAIL PROTECTED]
#
# This program is free software, distributed under the terms of
# the GNU General Public License
#
APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_mp3.so\
app_system.so app_echo.so app_record.so app_image.so app_url.so
app_disa.so \
app_adsiprog.so app_getcpeid.so app_milliwatt.so \
app_zapateller.so app_setcallerid.so app_festival.so \
app_queue.so app_senddtmf.so app_parkandannounce.so app_striplsd.so \
app_setcidname.so app_lookupcidname.so app_substring.so app_macro.so \
app_authenticate.so app_softhangup.so app_lookupblacklist.so \
app_waitforring.so app_privacy.so app_db.so app_chanisavail.so \
app_enumlookup.so app_transfer.so app_setcidnum.so app_cdr.so \
app_hasnewvoicemail.so app_sayunixtime.so app_cut.so app_read.so \
app_setcdruserfield.so app_random.so app_ices.so app_eval.so \
app_nbscat.so app_sendtext.so app_exec.so app_sms.so \
app_groupcount.so app_txtcidname.so app_controlplayback.so \
app_talkdetect.so app_alarmreceiver.so app_userevent.so app_verbose.so \
app_test.so app_forkcdr.so app_math.so app_realtime.so \
app_dumpchan.so app_waitforsilence.so app_while.so app_setrdnis.so \
app_md5.so app_readfile.so app_chanspy.so app_settransfercapability.so \
app_dictate.so app_externalivr.so app_directed_pickup.so app_rxfax.so
app_txfax.so
#
# Obsolete things...
#
#APPS+=app_sql_postgres.so
#APPS+=app_sql_odbc.so
#
# Experimental things
#
#APPS+=app_ivrdemo.so
#APPS+=app_skel.so
#APPS+=app_rpt.so
ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/linux/zaptel.h)$(wildcard
$(CROSS_COMPILE_TARGET)/usr/local/include/zaptel.h),)
APPS+=app_zapras.so app_meetme.so app_flash.so app_zapbarge.so app_zapscan.so
endif
ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/osp/osp.h),)
APPS+=app_osplookup.so
endif
ifeq ($(findstring BSD,${OSARCH}),BSD)
CFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include
-L$(CROSS_COMPILE_TARGET)/usr/local/lib
endif
CURLLIBS=$(shell $(CROSS_COMPILE_BIN)curl-config --libs)
ifneq ($(shell if [[ 0x`$(CROSS_COMPILE_BIN)curl-config --vernum` -ge 0x70907
]]; then echo OK ; fi),)
ifneq (${CURLLIBS},)
APPS+=app_curl.so
endif
endif
CFLAGS+=-fPIC
#
# If you have UnixODBC you can use ODBC voicemail
# storage
#
# Uncomment to use ODBC storage
#CFLAGS+=-DUSE_ODBC_STORAGE
# Uncomment for extended ODBC voicemail storage
#CFLAGS+=-DEXTENDED_ODBC_STORAGE
# See doc/README.odbcstorage for more information
all: $(APPS)
clean:
rm -f *.so *.o look .depend
%.so : %.o
$(CC) $(SOLINK) -o $@ $
app_rpt.so : app_rpt.o
$(CC) $(SOLINK) -o $@ $ -ltonezone
install: all
for x in $(APPS); do $(INSTALL) -m 755 $$x $(DESTDIR)$(MODULES_DIR) ;
done
rm -f $(DESTDIR)$(MODULES_DIR)/app_datetime.so
rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so
app_curl.so: app_curl.o
$(CC) $(SOLINK) -o $@ $ $(CURLLIBS)
app_rxfax.so : app_rxfax.o
$(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff
app_txfax.so : app_txfax.o
$(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff
app_sql_postgres.o: app_sql_postgres.c
$(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o
app_sql_postgres.o app_sql_postgres.c
app_sql_postgres.so: app_sql_postgres.o
$(CC) $(SOLINK) -o $@ $ -L/usr/local/pgsql/lib -lpq
app_sql_odbc.so: app_sql_odbc.o
$(CC) $(SOLINK) -o $@ $ -lodbc
look: look.c
$(CC) -pipe -O6 -g look.c -o look -lncurses
ifeq (SunOS,$(shell uname))
app_chanspy.so: app_chanspy.o
$(CC) $(SOLINK) -o $@ $ -lrt
endif
ifneq ($(wildcard .depend),)
include .depend
endif
depend: .depend
.depend:
../build_tools/mkdep $(CFLAGS) `ls *.c`
env:
env
rtxfax.c
I had to delete a routine that checked for chan_callerid in an old release,
And bingo it complied.
I do not know if it is stable howeer I will test over the weekend...
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
David Sampson
Sent: Friday, September 16, 2005 9:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] RxFax/TxFax - Compile Problem
If I understand correctly you are supposed to patch the
Makefile in the apps directory and then run the main
Makefile. I've tried both ways - the patch failed on the
main Makefile. Should I try to make that work?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[Asterisk-Users] alsa issue with asound.conf
I am using alsa with asterisk.
The asound.conf is below.
When I start asterisk with /etc/asound.conf present I get errors on
the console that:
chan_alsa.c:304 alsa_card_init: snd_pcm_open failed: Invalid argument
If I remove the asound.conf asterisk starts up and works. However
I NEED the asound.conf for another application.
What might be the issue here?
THanks
jerry
/etc/asound.conf
-
# This is /etc/asound.conf
#
# this makes legacy OSS apps use alsa software mixing dmix
pcm.dsp0 {
type plug
slave.pcm dmix
}
# mixer0 can stay unchanged, because it isn't used anyway, I guess ? ;)
ctl.mixer0 {
type hw
card 0
}
# this makes native ALSA apps default to using dmix
pcm.!default {
type plug
slave.pcm dmix
}
---
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[Asterisk-Users] queue_log on mysql
Hello, is there a best practice to upload queue_log file into MySQL? or - better - to have Asterisk log the queue_log straight to MySQL? Is it worth doing? Thanks l. -- Assum est, versa et manduca. ___ --Bandwidth and Colocation sponsored by Easynews.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] Asterisk don't start
Asterisk don't start, because show this message: Sep 16 17:04:59 WARNING[1075709024]: cli.c:702 ast_cli_register: Command 'showparkedcalls' already registered (or something close enough) == Parsing '/etc/asterisk/features.conf': Found -- Added extension '700' priority 1 to parkedcalls /usr/sbin/asterisk: relocation error: /usr/lib/asterisk/modules/res_features.so: undefined symbol: ast_pthread_create THANKS! ___ --Bandwidth and Colocation sponsored by Easynews.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] direct sip call pickup
Hi, im working about sip call pick and *8 works very fine but I pickup ringing phone on the same group. What happen if I have more than one ringing call? I tryied *8#exten, *8eten# but it doesnt wotk. Is it correct? How it does work ? Thanks Giordano ___ --Bandwidth and Colocation sponsored by Easynews.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] Caller Name: Asterisk reading too fast
As was already suggested, Wait() is your friend. We had the same problem when our PRI was installed. It was supposed to include Caller ID Name delivery, but it seemed to be hit-or-miss as to if it would work. This is what I found: When people call our auto-attendant and dial an extension, the name would work fine. However, if they dialed one of our DIDs (each extension has its own DID) then the name did not work. Inserting a Wait() command on the DIDs for extensions seemed to do the trick, like this: exten = 423303,1,Wait(0.75) exten = 423303,2,Dial(SIP/xxx) Here's my take on what happened: A call comes in to the auto-attendant, which immediately answers and begins the greeting. Shortly after, the telco sends the name down the PRI, and Asterisk associates it to that call. Then the caller punches in the extension and it goes through with name. (Keep in mind it only takes a fraction of a second for the name to show up on the PRI, so by the time a normal human realizes the auto-attendant has answered and dials an extension, plenty of time has passed for the name to come in.) However, the DIDs were spwaning Dial() *immediately* when the call came in. Of course, at that point the name was not present yet. (The telco was still doing the lookup.) But, Dial() has to send CID immediately when invoked, so it sent the number twice. Adding the 0.75 second delay gave time for the name to arrive before spawning Dial(). I've found 0.75 to be a fairly reliable delay. It misses a name here or there, but not often enough to be a problem. Most callers won't really notice the delay. If I increased this to one full second, it would be more noticeable but would probably fix the few ones we miss. You may have to play with this value to find the optimum setting for your setup. Another test you should be able to do: Have a call come in (when it shows only the number twice) and answer it. Then login to the Asterisk console, do show channels and find out which Zap channel it's on. Do a show channel Zap/whatever (whichever channel it is) and see if the name has shown up. IIRC, before I added the Wait, the show channel would still see the name, even though the phone didn't. Good luck! Jeremy J Thomas wrote: I asked my telco to release caller name on the PRI. Earlier they were releasing only the phone number. I still did not see the name, but only the number in caller id. Actually I now see number twice. When I inquired with them this is the response I got: I ran a trace on your TG. I see that your switch is picking up the call so fast that it is not able to pick up the name. The name is being sent, but I suspect after it is too late. This is something that will need to be corrected in your switch. I have attached a sample call out of the trace I performed this morning. They have sent me the trace file. Is there a way as it is in Asterisk so that it reads the caller name properly? Thanks, -- jt ___ --Bandwidth and Colocation sponsored by Easynews.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 -- Jeremy Gault, KD4NED[EMAIL PROTECTED] Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 msn msgr: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.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] direct sip call pickup
On CVS head there is app_directed_pickup It will let you pickup a ringing extension directly without having to worry about pickup groups etc. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giordano GrandisSent: Friday, September 16, 2005 11:19 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] direct sip call pickup Hi, im working about sip call pick and *8 works very fine but I pickup ringing phone on the same group. What happen if I have more than one ringing call? I tryied *8#exten, *8eten# but it doesnt wotk. Is it correct? How it does work ? Thanks Giordano ___ --Bandwidth and Colocation sponsored by Easynews.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] Caller Name: Asterisk reading too fast
This is because Caller ID name is being send in the FACILITY messages instead of in the SETUP. Wait(1) is the solution, There is nothing wrong with your install or asterisk. Asterisk picks up on SETUP as it should. When it recieves the Name via FACILITY it propigates it in the channel, however the other channel has already started to ring so it does not have the name when it gets built and since ZAP, IAX, and SIP, etc. do not support Caller ID after the setup you never get it Search on the Mantis for this -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Gault Sent: Friday, September 16, 2005 11:29 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Caller Name: Asterisk reading too fast As was already suggested, Wait() is your friend. We had the same problem when our PRI was installed. It was supposed to include Caller ID Name delivery, but it seemed to be hit-or-miss as to if it would work. This is what I found: When people call our auto-attendant and dial an extension, the name would work fine. However, if they dialed one of our DIDs (each extension has its own DID) then the name did not work. Inserting a Wait() command on the DIDs for extensions seemed to do the trick, like this: exten = 423303,1,Wait(0.75) exten = 423303,2,Dial(SIP/xxx) Here's my take on what happened: A call comes in to the auto-attendant, which immediately answers and begins the greeting. Shortly after, the telco sends the name down the PRI, and Asterisk associates it to that call. Then the caller punches in the extension and it goes through with name. (Keep in mind it only takes a fraction of a second for the name to show up on the PRI, so by the time a normal human realizes the auto-attendant has answered and dials an extension, plenty of time has passed for the name to come in.) However, the DIDs were spwaning Dial() *immediately* when the call came in. Of course, at that point the name was not present yet. (The telco was still doing the lookup.) But, Dial() has to send CID immediately when invoked, so it sent the number twice. Adding the 0.75 second delay gave time for the name to arrive before spawning Dial(). I've found 0.75 to be a fairly reliable delay. It misses a name here or there, but not often enough to be a problem. Most callers won't really notice the delay. If I increased this to one full second, it would be more noticeable but would probably fix the few ones we miss. You may have to play with this value to find the optimum setting for your setup. Another test you should be able to do: Have a call come in (when it shows only the number twice) and answer it. Then login to the Asterisk console, do show channels and find out which Zap channel it's on. Do a show channel Zap/whatever (whichever channel it is) and see if the name has shown up. IIRC, before I added the Wait, the show channel would still see the name, even though the phone didn't. Good luck! Jeremy J Thomas wrote: I asked my telco to release caller name on the PRI. Earlier they were releasing only the phone number. I still did not see the name, but only the number in caller id. Actually I now see number twice. When I inquired with them this is the response I got: I ran a trace on your TG. I see that your switch is picking up the call so fast that it is not able to pick up the name. The name is being sent, but I suspect after it is too late. This is something that will need to be corrected in your switch. I have attached a sample call out of the trace I performed this morning. They have sent me the trace file. Is there a way as it is in Asterisk so that it reads the caller name properly? Thanks, -- jt ___ --Bandwidth and Colocation sponsored by Easynews.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 -- Jeremy Gault, KD4NED[EMAIL PROTECTED] Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 msn msgr: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing
[Asterisk-Users] mpg123 on x86_64 (Opteron MP)
Has anyone successfully compiled mpg123 in the 1.0.x or 1.2beta1 distributions (I'm running FC3 linux on an Opteron 2 processor system)? Are there any patches out there to make it work? gcc -DI386_ASSEM -DPENTIUM_OPT -DREAL_IS_FLOAT -DLINUX - DREAD_MMAP -DOSS -DTERM_CONTROL-Wall -O2 - m486 -fomit-frame-pointer -funroll-all- loops -finline-functions -ffast-math -c -o dct64_i386.o dct64_i386.c as -o decode_i586.o decode_i586.s decode_i586.s: Assembler messages: decode_i586.s:44: Error: suffix or operands invalid for `push' snip decode_i586.s:161: Error: suffix or operands invalid for `pop' snip ___ --Bandwidth and Colocation sponsored by Easynews.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 version of idefisk softphone released.
We just uploaded the latest and greatest version of the idefisk iax2 softphone, version 1.24 Freely downloadable at: http://www.asteriskguru.com/tools/idefisk_beta.php Changes since the last release include: - history panel is working - receiving messages and urls (sendtext command in asterisk) - some bugfixes (the annoying hangup bug is finally gone!). A big thanks to everybody who sent us comments and suggestions ! Zoa signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation sponsored by Easynews.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] Caller Name: Asterisk reading too fast
On Fri, 16 Sep 2005, J Thomas wrote: I asked my telco to release caller name on the PRI. Earlier they were releasing only the phone number. I still did not see the name, but only the number in caller id. Actually I now see number twice. When I inquired with them this is the response I got: I ran a trace on your TG. I see that your switch is picking up the call so fast that it is not able to pick up the name. The name is being sent, but I suspect after it is too late. This is something that will need to be corrected in your switch. I have attached a sample call out of the trace I performed this morning. They have sent me the trace file. Is there a way as it is in Asterisk so that it reads the caller name properly? Sheesh - you should really look after that support contact at your telco. Than ran a trace, looked at it, understood it and gave you great advice on how to solve the problem. That must be 1 in a 1000 support! Steve ___ --Bandwidth and Colocation sponsored by Easynews.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
R: [Asterisk-Users] direct sip call pickup
I cannot use CVS, is there anoyher way to use direct pickup ? Thanks again Giordano Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Alexander Lopez Inviato: venerdì 16 settembre 2005 17.53 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: RE: [Asterisk-Users] direct sip call pickup On CVS head there is app_directed_pickup It will let you pickup a ringing extension directly without having to worry about pickup groups etc. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis Sent: Friday, September 16, 2005 11:19 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] direct sip call pickup Hi, im working about sip call pick and *8 works very fine but I pickup ringing phone on the same group. What happen if I have more than one ringing call? I tryied *8#exten, *8eten# but it doesnt wotk. Is it correct? How it does work ? Thanks Giordano ___ --Bandwidth and Colocation sponsored by Easynews.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: R: [Asterisk-Users] direct sip call pickup
see if it compiles into ur install Iqbal Giordano Grandis wrote: I cannot use CVS, is there anoyher way to use direct pickup ? Thanks again **Giordano** *Da:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *Per conto di *Alexander Lopez *Inviato:* venerdì 16 settembre 2005 17.53 *A:* Asterisk Users Mailing List - Non-Commercial Discussion *Oggetto:* RE: [Asterisk-Users] direct sip call pickup On CVS head there is app_directed_pickup It will let you pickup a ringing extension directly without having to worry about pickup groups etc. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Giordano Grandis *Sent:* Friday, September 16, 2005 11:19 AM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] direct sip call pickup Hi, i’m working about sip call pick and *8 works very fine but I pickup ringing phone on the same group. What happen if I have more than one ringing call? I tryied *8#exten, *8eten# but it doesn’t wotk. Is it correct? How it does work ? Thanks **Giordano** ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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: use of automon sequence (was Re: [Asterisk-Users] Call recording between SIP phones)
Thanks -- it works very nicely -- I will have to try that filename, but otherwise it seems to be just fine. on Friday 09/16/2005 Kevin Bockman([EMAIL PROTECTED]) wrote John covici wrote: OK, I wonder if I have something wrong -- I have the *1 in my features.conf for the one touch record -- now I called a number, and when the call was answered flashed the hook and pressed *1 and went back tothe call, but nothing happpened. I am using CVS from 8/26 -- is this too old or am I doing something wrong? I did add wW to the dial options of the call and that part did work, but there is nothing in the log about automon. Also, what kind of file does this produce, etc? Thanks in advance for any help. I'm not sure when it started, but as long as you have it in the example features.conf in your asterisk source/configs/features.conf file. You don't do a flash to do the record. You press the sequence WHILE YOU ARE ON THE CALL. It will record a file to /var/spool/monitor. To change the filename, you probably have to set the MONITOR_FILENAME variable before the dial. Kevin ___ --Bandwidth and Colocation sponsored by Easynews.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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.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] Asterisk as a gateway. 'flash for transfers transparency?'
Hi,
I have 2 asterisk boxes as Gateway, in this arrangement.
(PANASONIC PBX) - [ASTERISK1] - network - [ASTERISK2] - (ANALOG PHONE)
everything works great, in both directions (receiving and making calls),
but when i get a call on the (ANALOGPHONE), I haven't been able to
transfer it to another extension of the PANASONIC PBX using the flash key.
I've tried the using the t T options on the 'Dial(' command of the extensions, but no
luck. I assume that's more for transfering on the asterisk domain. I just want to pass
the flash key through the 2 asterisk boxes to the panasonic PBX. (Call transfer, blind i
would say).
anyone have some insight on how to do this? ( i haven't been able to locate any documentation
on the web)
___
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RE: [Asterisk-Users] RxFax/TxFax - Compile Problem
On Fri, 2005-09-16 at 09:36 -0400, David Sampson wrote: If I understand correctly you are supposed to patch the Makefile in the apps directory and then run the main Makefile. I've tried both ways - the patch failed on the main Makefile. Should I try to make that work? There's next to nothing in the patchfile so edit the Makefile manually. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.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] queue_log on mysql
Best to log directly to MySQL. Add in ODBC code. You are not the first to ask for it. -bill On 16-Sep-05, at 11:06 AM, lenz wrote: Hello, is there a best practice to upload queue_log file into MySQL? or - better - to have Asterisk log the queue_log straight to MySQL? Is it worth doing? Thanks l. -- Assum est, versa et manduca. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: T.38 ATA
PA-122TI from www.broad-tel.com supports T.38 and Fax pass-thru. On 9/15/05, Rosario Pingaro [EMAIL PROTECTED] wrote: about spa-2100, the t38 stream is on UDPTL and so asterisk passthroughdoesn't work.- Original Message - From: Nenad Radosavljevic [EMAIL PROTECTED]To: asterisk-users@lists.digium.comSent: Wednesday, September 14, 2005 12:16 PM Subject: [Asterisk-Users] Re: T.38 ATA Hi ! First of all thank you all for fast response on matter of T.38 capable ATAs. I have asked a UK VoIP suplier to check with manufacterers of various ATAs they sell, do they support T.38 and here is what they/I have got as a result: 1. Sipura SPA-2100 only and with firmware 3.2.1 is T.38 capable (no information on type of T.28 support UDPTL/TPKT) 2. All Gradnstream Handytone ATAs with firmware grater than 1.0.6.x are T.38 capable and they use UDPTL T.38 Regards,Nenad The newest 2100 firmware has T.38. ___ --Bandwidth and Colocation sponsored by Easynews.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___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.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] Easier way for end user to change main greeting?
Hi, Has someone figured out how to change the main autoattendant message easily? Right now, you call *77 and record the message. Then you have to get into the Unix/Linux command line to get that message over to where it will be used. Is there a simpler way? Thanks for your help. ___ --Bandwidth and Colocation sponsored by Easynews.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] RxFax/TxFax - Compile Problem
Maybe I should ask this question that I know has been discussed to death. "stable" = 1.0 release "CVS HEAD' = 1.1 release Is this a correct statment From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David SampsonSent: Thursday, September 15, 2005 12:17 PMTo: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] RxFax/TxFax - Compile Problem I used the latest version (.3) and also the previous .2 ver (pre20). The spandsp seems to compile but when I download the rxfax/txfax .c files and drop them in the apps directory that is where I get the compile error. Dave From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ WeschkeSent: Wednesday, September 14, 2005 2:43 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] RxFax/TxFax - Compile Problem What version of spandsp are you attempting to compile in to the 1.0.9 tree? On 9/14/05, David Sampson [EMAIL PROTECTED] wrote: Anyone know how to fix this?gcc -shared -Xlinker -x -o app_rxfax.so app_rxfax.c -lspandsp -ltiff In file included from app_rxfax.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:302: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) /usr/include/asterisk/lock.h:302: error: (Each undeclared identifier is reported only once /usr/include/asterisk/lock.h:302: error: for each function it appears in.) app_rxfax.c: In function `rxfax_exec': app_rxfax.c:263: warning: passing arg 1 of `fax_init' from incompatible pointer type app_rxfax.c:264: error: structure has no member named `verbose' app_rxfax.c:325: warning: passing arg 1 of `fax_release' from incompatible pointer type make[1]: *** [app_rxfax.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.0.9/apps' make: *** [subdirs] Error 1 ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.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: R: [Asterisk-Users] direct sip call pickup
What you were trying to do, *8#exten, is almost right I think. Look at it like this instead, though. The # is a pickup group number: *8x where x is the pickup group you want to pick up a call from. I could be wrong but that's how I understood it. Mojo Giordano Grandis wrote: I cannot use CVS, is there anoyher way to use direct pickup ? Thanks again **Giordano** *Da:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *Per conto di *Alexander Lopez *Inviato:* venerdì 16 settembre 2005 17.53 *A:* Asterisk Users Mailing List - Non-Commercial Discussion *Oggetto:* RE: [Asterisk-Users] direct sip call pickup On CVS head there is app_directed_pickup It will let you pickup a ringing extension directly without having to worry about pickup groups etc. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Giordano Grandis *Sent:* Friday, September 16, 2005 11:19 AM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] direct sip call pickup Hi, i’m working about sip call pick and *8 works very fine but I pickup ringing phone on the same group. What happen if I have more than one ringing call? I tryied *8#exten, *8eten# but it doesn’t wotk. Is it correct? How it does work ? Thanks **Giordano** ___ --Bandwidth and Colocation sponsored by Easynews.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 -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation sponsored by Easynews.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] Sipura 2k voice quality
When I have voip conversations over asterisk through my computer the voice quality is nice and loud and quite clear. When I go through my Sipura 2K then the conversations are typically very muted and my responses sound somewhat delayed. I've tried fiddling with settings under asterisk and the 2K but haven't had any luck. Can someone point me in the right direction to get this configured right? -- Matthew Harrell Beauty is in the eye of the beer Bit Twiddlers, Inc. holder. [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.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] Zap failed
I have configured /etc/asterisk/zapata.conf, but now Asterisk refuses to start: Asterisk 1.0.9, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Sep 16 20:36:51 ERROR[6750]: chan_zap.c:6246 mkintf: Unable to get parameters Sep 16 20:36:51 ERROR[6750]: chan_zap.c:9191 setup_zap: Unable to register channel '1-15' Sep 16 20:36:51 WARNING[6750]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Sep 16 20:36:51 WARNING[6750]: loader.c:440 load_modules: Loading module chan_zap.so failed! sip:/etc # Ouch ... error while writing audio data: : Broken pipe Here is the /proc/zaptel/1 file(seems to be correct, and log Messages too indicates the initialization is correct.): cat /proc/zaptel/1 Span 1: WCT1/0 Digium Wildcard E100P E1/PRA Card 0 HDB3/CCS ClockSource 1 WCT1/0/1 Clear 2 WCT1/0/2 Clear 3 WCT1/0/3 Clear 4 WCT1/0/4 Clear 5 WCT1/0/5 Clear 6 WCT1/0/6 Clear 7 WCT1/0/7 Clear 8 WCT1/0/8 Clear 9 WCT1/0/9 Clear 10 WCT1/0/10 Clear 11 WCT1/0/11 Clear 12 WCT1/0/12 Clear 13 WCT1/0/13 Clear 14 WCT1/0/14 Clear 15 WCT1/0/15 Clear 16 WCT1/0/16 HDLCFCS 17 WCT1/0/17 Clear 18 WCT1/0/18 Clear 19 WCT1/0/19 Clear 20 WCT1/0/20 Clear 21 WCT1/0/21 Clear 22 WCT1/0/22 Clear 23 WCT1/0/23 Clear 24 WCT1/0/24 Clear 25 WCT1/0/25 Clear 26 WCT1/0/26 Clear 27 WCT1/0/27 Clear 28 WCT1/0/28 Clear 29 WCT1/0/29 Clear 30 WCT1/0/30 Clear 31 WCT1/0/31 Clear Ztcfg -vv Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: Individual Clear channel (Default) (Slaves: 24) Channel 25: Individual Clear channel (Default) (Slaves: 25) Channel 26: Individual Clear channel (Default) (Slaves: 26) Channel 27: Individual Clear channel (Default) (Slaves: 27) Channel 28: Individual Clear channel (Default) (Slaves: 28) Channel 29: Individual Clear channel (Default) (Slaves: 29) Channel 30: Individual Clear channel (Default) (Slaves: 30) Channel 31: Individual Clear channel (Default) (Slaves: 31) 31 channels configured. /etc/zaptel file: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=us defaultzone=us /etc/asterisk/zapata.conf [channels] switchtype = euroisdn signalling = pri_net pridialplan = local language=en context=ivr-in overlapdial=yes usecallerid=yes hidecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no musiconhold=default group=1 channel = 1-15 channel = 17-31 smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation sponsored by Easynews.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] wiki down?
I'm unable to connect to voip-info.org... Anybody else have the same issues, ro is it just me? -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.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] didn't get a frame from channel
This is an excerpt from the log file, My problem is that randomly, 1out of 3 or 1 out of 2, some calls are not going out and this is the message in the log file, The device that should provide the frame is a Sipura 3000 which has its FXO providing outside connectivity, 24185 Sep 16 10:35:40 DEBUG[17604]: Didn't get a frame from channel: SIP/200-b635 24186 Sep 16 10:35:40 DEBUG[17604]: Bridge stops bridging channels SIP/200-b635 and SIP/pstn_1-a971 24187 Sep 16 10:35:40 DEBUG[17604]: update_user_counter(ww9863038) - decrement outUse counter 24188 Sep 16 10:35:40 DEBUG[17604]: ww9863038 is not a local user 24189 Sep 16 10:35:40 DEBUG[17604]: Exiting with DIALSTATUS=ANSWER. I can provide the whole call log but it's many lines long. Any clue on where to look for anything? Thanks Andres ___ --Bandwidth and Colocation sponsored by Easynews.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] wiki down?
I got right in just fine... W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francesco Peeters Sent: Friday, September 16, 2005 10:26 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] wiki down? I'm unable to connect to voip-info.org... Anybody else have the same issues, ro is it just me? -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Sipura 2k voice quality
Raise both gains from -3 to 5 that solves volume problem, log in, click admin, advanced, I guess is on the sip tab, Matthew Harrell wrote: When I have voip conversations over asterisk through my computer the voice quality is nice and loud and quite clear. When I go through my Sipura 2K then the conversations are typically very muted and my responses sound somewhat delayed. I've tried fiddling with settings under asterisk and the 2K but haven't had any luck. Can someone point me in the right direction to get this configured right? ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: Polycom randomly fails outbound calls,
Noah Miller wrote: Hi Andres - The two that we have are just used as lobby phones. They're good little phones, but if you have the money, I'd definitely recommend the IP501 instead. The screen is MUCH better, and having full speakerphone is great! Plus the 500/501 just feels a little more solid. Yeah, I think it was a wrong move going for the 301 instead, Hmm. I'm not sure either. I've never used AMP before (except for a quick glance at [EMAIL PROTECTED]). If you can change the sip settings, I don't think it should matter. I am not using [EMAIL PROTECTED], it didn't work well for me, I just using AMP over asterisk, and yes, sip are 100% tweakable, how do you configure your system, all by hand? Well, the two weird things I see here are the type setting and the host. Type is set to peer, but there doesn't seem to be a corresponding user definition (AFAIK, all peers have to have users). You might try changing it to type=friend instead (like 201). I did it, it was set to peer just because I red somewhere that Polys didn't like friend type, For the host setting, this is the address of the sip device, and not the asterisk server. If you have the Polycom set to a static address of 192.168.1.18, all is well. If your Polycom is set to DHCP (this is the default), you should use host=dynamic it's fixed to 18 A couple of things that I know you don't need: nat=never qualify=no I took them off too, I got them from the only how to I found about amp and the polycom, - Noah Thanks , I hope I can help you same day, Andres ___ --Bandwidth and Colocation sponsored by Easynews.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] lastest spandsp-0.03pre1 don't compile
Dear all, Anyone get the lastest spandsp with udptl.c and tpkt.c compile in Fedora 3? tpkt.c: In function `accept_thread': tpkt.c:140: error: `TCP_NODELAY' undeclared (first use in this function) tpkt.c:140: error: (Each undeclared identifier is reported only once tpkt.c:140: error: for each function it appears in.) tpkt.c:144: error: invalid application of `sizeof' to incomplete type `mansession' tpkt.c:148: error: invalid application of `sizeof' to incomplete type `mansession' tpkt.c:149: error: dereferencing pointer to incomplete type tpkt.c:151: error: `block_sockets' undeclared (first use in this function) tpkt.c:156: error: dereferencing pointer to incomplete type tpkt.c:157: error: dereferencing pointer to incomplete type tpkt.c:158: error: dereferencing pointer to incomplete type tpkt.c:159: error: `sessionlock' undeclared (first use in this function) tpkt.c:160: error: dereferencing pointer to incomplete type tpkt.c:160: error: `sessions' undeclared (first use in this function) tpkt.c:163: error: `t' undeclared (first use in this function) tpkt.c:163: error: `session_do' undeclared (first use in this function) tpkt.c:164: warning: implicit declaration of function `destroy_session' tpkt.c: At top level: tpkt.c:171: warning: no previous prototype for 'init_tpkt' tpkt.c: In function `init_tpkt': tpkt.c:173: warning: passing arg 1 of `pthread_kill' makes integer from pointer without a cast tpkt.c:177: error: `addr' undeclared (first use in this function) tpkt.c:178: error: `portno' undeclared (first use in this function) tpkt.c:179: error: `ba' undeclared (first use in this function) tpkt.c:180: error: `val' undeclared (first use in this function) tpkt.c:180: error: `cfg' undeclared (first use in this function) tpkt.c:188: warning: `return' with a value, in function returning void tpkt.c:190: error: `x' undeclared (first use in this function) tpkt.c:195: warning: `return' with a value, in function returning void tpkt.c:201: warning: `return' with a value, in function returning void tpkt.c:205: error: `t' undeclared (first use in this function) tpkt.c: In function `tpkt_rx_packet': tpkt.c:241: warning: implicit declaration of function `decode_open_type' tpkt.c:211: warning: unused variable `stat2' tpkt.c:212: warning: unused variable `i' tpkt.c:213: warning: unused variable `j' udptl.c: In function `udptl_process_packet': udptl.c:150: warning: no return statement in function returning non-void udptl.c: In function `udptl_build_packet': udptl.c:541: warning: implicit declaration of function `udptl_debug_test_addr' udptl.c:541: error: `udptl' undeclared (first use in this function) udptl.c:541: error: (Each undeclared identifier is reported only once udptl.c:541: error: for each function it appears in.) udptl.c:543: error: `iabuf' undeclared (first use in this function) udptl.c:544: error: `payload' undeclared (first use in this function) udptl.c:544: error: `res' undeclared (first use in this function) udptl.c:544: error: `hdrlen' undeclared (first use in this function) udptl.c: At top level: udptl.c:571: warning: static declaration of 'udptl_debug_test_addr' follows non-static declaration udptl.c:541: warning: 'udptl_debug_test_addr' declared inline after being called udptl.c:541: warning: previous implicit declaration of 'udptl_debug_test_addr' was here udptl.c: In function `ast_udptl_read': udptl.c:643: error: `payloadtype' undeclared (first use in this function) udptl.c:643: error: `timestamp' undeclared (first use in this function) udptl.c:643: error: `hdrlen' undeclared (first use in this function) udptl.c:649: error: `AST_FORMAT_T38' undeclared (first use in this function) udptl.c: In function `ast_udptl_new_with_bindaddr': udptl.c:706: error: `s' undeclared (first use in this function) udptl.c: In function `ast_udptl_write': udptl.c:820: error: `codec' undeclared (first use in this function) udptl.c:820: error: structure has no member named `lastts' udptl.c:820: error: `hdrlen' undeclared (first use in this function) udptl.c: In function `ast_udptl_reload': udptl.c:1084: warning: implicit declaration of function `ast_load' udptl.c:1084: warning: assignment makes pointer from integer without a cast udptl.c:1138: warning: implicit declaration of function `ast_destroy' ray ___ --Bandwidth and Colocation sponsored by Easynews.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] USB Phones for use with Asterisk
On Fri, 2005-09-16 at 11:54 +0100, c waddy wrote: Are there drivers available for Xten Softphones? Do they work with X-lite/Eyebeam Softphones? And work on Linux (kernel 2.6)? On 9/16/05, Bill McCready (PCPhoneline.com) [EMAIL PROTECTED] wrote: Hi all, I have a question that I was hoping someone could answer for me. I would like to find a USB phone that works with Asterisk... preferably one that does not have any issues with NAT. RESPONSE: This is Bill McCready from PCPhoneline.com . I suggest you give our VTA1000 Skype+SIP+IAX2 Gateway which is USB based with a good cordless phone. We offer several versions of the VTA1000 but this is the latest one we have developed and it supports Skype + 10 x SIP + 10 x IAX2 networks all at the same time and plugs into the USB port of a Windows 2000 or Windows XP computer. You can optionally disable Skype and/or the SIP calling capabilities if you don't need them. The VTA1000 is designed to work even if your computer screen is powered off using it with a cordless phone of course gives you the added advantage of not needing to be tied to your computer to place and receive calls. If you don't mind being tethered to a computer we also have the VPT1000 phone available with the same capability set. You can find them in our Ebay store at http://stores.ebay.com/pcphoneline . If you need a bunch of these and want a build specially tailored and branded please private e-mail me with your requirements and we will do our best to try to accomodate you. Best regards...Bill Can anyone point me to something suitable ? We are essentially planning to do a beta for a few hundred customers on our network and would like them to be able to call one another @ no charge. Because these are going to be end users who have no idea about Voice / VOIP stuff... it needs to be extremely easy to use. Can someone point me in the right direction ? Cheers, Callum ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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 -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --Bandwidth and Colocation sponsored by Easynews.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] queue_log on mysql
Thanks, is there a standard schema for queue_log or can I define it myself? Thanks l. In data Fri, 16 Sep 2005 18:48:13 +0200, William Lloyd [EMAIL PROTECTED] ha scritto: Best to log directly to MySQL. Add in ODBC code. You are not the first to ask for it. -bill On 16-Sep-05, at 11:06 AM, lenz wrote: Hello, is there a best practice to upload queue_log file into MySQL? or - better - to have Asterisk log the queue_log straight to MySQL? Is it worth doing? Thanks l. --Assum est, versa et manduca. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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 -- Assum est, versa et manduca. ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: Polycom randomly fails outbound calls,
Hi Andres - I am not using [EMAIL PROTECTED], it didn't work well for me, I just using AMP over asterisk, and yes, sip are 100% tweakable, how do you configure your system, all by hand? Yeah, by hand. When I first started doing this there was no such thing as AMP. Plus, I've got some wacky dialplan stuff that probably wouldn't work out too well with AMP. Well, the two weird things I see here are the type setting and the host. Type is set to peer, but there doesn't seem to be a corresponding user definition (AFAIK, all peers have to have users). You might try changing it to type=friend instead (like 201). I did it, it was set to peer just because I red somewhere that Polys didn't like friend type, For the host setting, this is the address of the sip device, and not the asterisk server. If you have the Polycom set to a static address of 192.168.1.18, all is well. If your Polycom is set to DHCP (this is the default), you should use host=dynamic it's fixed to 18 Well, if you've got all this stuff and the phones still aren't working, I'd say there's either something funny going on with AMP or Asterisk. I would try new/different versions of both. The rest of your configs look good to me. Maybe somebody else can spot an inconsistency? Just to cover all the bases, have you tried configuring the Polycom without AMP? - Noah ___ --Bandwidth and Colocation sponsored by Easynews.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: R: [Asterisk-Users] direct sip call pickup
Hello Everyone, For regular call pickup you can't really specify a pickup group number... that's why it's set in the configuration. For directed call pickup you need to have the latest CVS head as it uses an API call that Kevin put in espically for me to use lastnight. Joshua Colp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Friday, September 16, 2005 2:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: R: [Asterisk-Users] direct sip call pickup What you were trying to do, *8#exten, is almost right I think. Look at it like this instead, though. The # is a pickup group number: *8x where x is the pickup group you want to pick up a call from. I could be wrong but that's how I understood it. Mojo Giordano Grandis wrote: I cannot use CVS, is there anoyher way to use direct pickup ? Thanks again **Giordano** *Da:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *Per conto di *Alexander Lopez *Inviato:* venerdì 16 settembre 2005 17.53 *A:* Asterisk Users Mailing List - Non-Commercial Discussion *Oggetto:* RE: [Asterisk-Users] direct sip call pickup On CVS head there is app_directed_pickup It will let you pickup a ringing extension directly without having to worry about pickup groups etc. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Giordano Grandis *Sent:* Friday, September 16, 2005 11:19 AM *To:* asterisk-users@lists.digium.com *Subject:* [Asterisk-Users] direct sip call pickup Hi, im working about sip call pick and *8 works very fine but I pickup ringing phone on the same group. What happen if I have more than one ringing call? I tryied *8#exten, *8eten# but it doesnt wotk. Is it correct? How it does work ? Thanks **Giordano** ___ --Bandwidth and Colocation sponsored by Easynews.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 -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Weird behaviour
Hi, I noticed this weird behavior - in my office I use mixed phone technology. I use Sip and Zap phones, analog and ISDN. I also defined a pickup feature and everything works prima to the time when I want to pickup call with ISDN phone. The console says (when I press my pickup extension *6) no such extension. Why? Other phones do not need special definition in extension.conf. When I added extra line in my dialplan - exten = *6,1,Pickup(group) - the ISDN phone was able to pickup all channels instead of Sip. To be certain the Sip channel was picked up, but the phone did not stop to ring. All phones are in the same pickup and call group. Do I need something special in my zapata.conf to normally pickup all channels or do I have to add some application to asterisk? Or I just did something wrong? Regards Andrutto -- Prawie 40.000 samochodow na sprzedaz! http://link.interia.pl/f18b2 ___ --Bandwidth and Colocation sponsored by Easynews.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] asterisk mixing sound card with anybody?
I am trying to get asterisk to MIX micely with alsa or oss. Anybody doing that? if so can you share with me how you did it. When I set modules.conf to load alsa and noload oss asterisk starts and binds the sound port. nothing else will play. If I stop asterisk other things play. If I put a file /etc/asound.conf asterisk doesnt like it (tried a couple ones) and will not use the sound port at all. This same config file allows other applications to mix nicely like mplayer and aplay at the same time. I dont want to have to stop asterisk to play a mplayer file or an aplay file. I then tried noload on alsa and load on oss. Pretty much with the same results. Can get anything else to play if asterisk is playing.. You help is appreciated. Thanks, Jerry ___ --Bandwidth and Colocation sponsored by Easynews.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] How to access * thru router when ip address is not known
I am going to be traveling and I wanted to be able to get on the internet and call thru * to make calls. The problem is I do not have a fixed ip address. How do you make this work? I will be using IAXCOMM. TIA Bob ___ --Bandwidth and Colocation sponsored by Easynews.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] How to access * thru router when ip address is not known
dyndns.org From: Robert Rawlinson [EMAIL PROTECTED] Date: 2005/09/16 Fri PM 03:51:56 EDT To: Asterisk asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to access * thru router when ip address is not known I am going to be traveling and I wanted to be able to get on the internet and call thru * to make calls. The problem is I do not have a fixed ip address. How do you make this work? I will be using IAXCOMM. TIA Bob ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: g729 to asterisk to g729 voip provider
anyone with some info on this? thanks again. On 9/14/05, Erick Perez [EMAIL PROTECTED] wrote: Using sipura sip/g729 to connect to an asterisk server that will server as a gateway to a VOIP provider, all in g729 will require to purchase codecs from Digium? also, in this scenario the transcoding is almost non-existent right? I have read many documents about the type of codecs, and g729 seems to be a good trade between almost-toll quality and low bandwith usage right? A P4 (HT) 3.2 Ghz with 1gb ram in a 800mhz board with 3 raid0 disk can sustain more than 100 calls or up to a 100? I just looking at hardware capacity, since the machine will be located at an ISP with more than needed bandwith. There is no need for voicemail, web interfaces or anything else, since the * box will only function as a gateway to a US-based VOIP provider. The machine in question runs Centos4 Linux (Redhat enterprise 4) and CDR logging only. Thanks, -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation sponsored by Easynews.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] wiki down?
On Fri, September 16, 2005 19:53, Wiley Siler said: I got right in just fine... W Me too now. :-/ -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: g729 to asterisk to g729 voip provider
Hello Erik! check out this website: http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ i have both codecs running on gentoo without problems(working with cisco 7960 and snom 190). regarding your hardware question i don't have enough experience yet, sorry. regards christian On Fri, 16 Sep 2005 15:01:32 -0500 Erick Perez [EMAIL PROTECTED] wrote: anyone with some info on this? thanks again. On 9/14/05, Erick Perez [EMAIL PROTECTED] wrote: Using sipura sip/g729 to connect to an asterisk server that will server as a gateway to a VOIP provider, all in g729 will require to purchase codecs from Digium? also, in this scenario the transcoding is almost non-existent right? I have read many documents about the type of codecs, and g729 seems to be a good trade between almost-toll quality and low bandwith usage right? A P4 (HT) 3.2 Ghz with 1gb ram in a 800mhz board with 3 raid0 disk can sustain more than 100 calls or up to a 100? I just looking at hardware capacity, since the machine will be located at an ISP with more than needed bandwith. There is no need for voicemail, web interfaces or anything else, since the * box will only function as a gateway to a US-based VOIP provider. The machine in question runs Centos4 Linux (Redhat enterprise 4) and CDR logging only. Thanks, -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] wiki down?
voip-info is down from time to time, guess more spending for their server hardware is needed. On Fri, 16 Sep 2005 19:26:18 +0200 (CEST) Francesco Peeters [EMAIL PROTECTED] wrote: I'm unable to connect to voip-info.org... Anybody else have the same issues, ro is it just me? -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] How to access * thru router when ip address is not known
Assuming your Asterisk mail client is set up correctly, cron this twice a day on your Asterisk box: ifconfig eth0 | grep inet addr | mail -s My Asterisk IP address [EMAIL PROTECTED] Check your mail before you call and verify that the IP address has not changed, if it has, modify your client accordingly. -Original Message- From: Robert Rawlinson [mailto:[EMAIL PROTECTED] Sent: Friday, September 16, 2005 1:52 PM To: Asterisk Subject: [Asterisk-Users] How to access * thru router when ip address is not known I am going to be traveling and I wanted to be able to get on the internet and call thru * to make calls. The problem is I do not have a fixed ip address. How do you make this work? I will be using IAXCOMM. TIA Bob ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] USB Phones for use with Asterisk
There is a standalone linux version of xlite available on their homepage.. http://www.xten.com On Fri, 2005-09-16 at 11:38, Guillermo Salas M wrote: On Fri, 2005-09-16 at 11:54 +0100, c waddy wrote: Are there drivers available for Xten Softphones? Do they work with X-lite/Eyebeam Softphones? And work on Linux (kernel 2.6)? On 9/16/05, Bill McCready (PCPhoneline.com) [EMAIL PROTECTED] wrote: Hi all, I have a question that I was hoping someone could answer for me. I would like to find a USB phone that works with Asterisk... preferably one that does not have any issues with NAT. RESPONSE: This is Bill McCready from PCPhoneline.com . I suggest you give our VTA1000 Skype+SIP+IAX2 Gateway which is USB based with a good cordless phone. We offer several versions of the VTA1000 but this is the latest one we have developed and it supports Skype + 10 x SIP + 10 x IAX2 networks all at the same time and plugs into the USB port of a Windows 2000 or Windows XP computer. You can optionally disable Skype and/or the SIP calling capabilities if you don't need them. The VTA1000 is designed to work even if your computer screen is powered off using it with a cordless phone of course gives you the added advantage of not needing to be tied to your computer to place and receive calls. If you don't mind being tethered to a computer we also have the VPT1000 phone available with the same capability set. You can find them in our Ebay store at http://stores.ebay.com/pcphoneline . If you need a bunch of these and want a build specially tailored and branded please private e-mail me with your requirements and we will do our best to try to accomodate you. Best regards...Bill Can anyone point me to something suitable ? We are essentially planning to do a beta for a few hundred customers on our network and would like them to be able to call one another @ no charge. Because these are going to be end users who have no idea about Voice / VOIP stuff... it needs to be extremely easy to use. Can someone point me in the right direction ? Cheers, Callum ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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 -- -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK-- signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.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] How to access * thru router when ip address is not known
whoops, should have prefaced that with the Asterisk box has to be forward of any firewall, otherwise it's going to return a 10.X.X.X or 192.168.X.X IP address. -Original Message- From: Colin Anderson [mailto:[EMAIL PROTECTED] Sent: Friday, September 16, 2005 2:28 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How to access * thru router when ip address is not known Assuming your Asterisk mail client is set up correctly, cron this twice a day on your Asterisk box: ifconfig eth0 | grep inet addr | mail -s My Asterisk IP address [EMAIL PROTECTED] Check your mail before you call and verify that the IP address has not changed, if it has, modify your client accordingly. -Original Message- From: Robert Rawlinson [mailto:[EMAIL PROTECTED] Sent: Friday, September 16, 2005 1:52 PM To: Asterisk Subject: [Asterisk-Users] How to access * thru router when ip address is not known I am going to be traveling and I wanted to be able to get on the internet and call thru * to make calls. The problem is I do not have a fixed ip address. How do you make this work? I will be using IAXCOMM. TIA Bob ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] Anyone using iPlan Networks in Argentina?
Hello, Is anyone successfully working with iPlan Networks in Argentina for telephony service? I'm interested in hearing about people's experience with their service and support. Regards, Ilan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: g729 to asterisk to g729 voip provider
Hi, your project is indeed interesting, however for learning purposes i do need to know the answer of at least: 1- Using sipura sip/g729 to connect to an asterisk server that will server as a gateway to a VOIP provider(g729), all in g729 will require to purchase codecs from Digium? 2- also, in this scenario the transcoding is almost non-existent right? 3- I have read many documents about the type of codecs, and g729 seems to be a good trade between almost-toll quality and low bandwith usage right? On 9/16/05, ChB [EMAIL PROTECTED] wrote: Hello Erik! check out this website: http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ i have both codecs running on gentoo without problems(working with cisco 7960 and snom 190). regarding your hardware question i don't have enough experience yet, sorry. regards christian On Fri, 16 Sep 2005 15:01:32 -0500 Erick Perez [EMAIL PROTECTED] wrote: anyone with some info on this? thanks again. On 9/14/05, Erick Perez [EMAIL PROTECTED] wrote: Using sipura sip/g729 to connect to an asterisk server that will server as a gateway to a VOIP provider, all in g729 will require to purchase codecs from Digium? also, in this scenario the transcoding is almost non-existent right? I have read many documents about the type of codecs, and g729 seems to be a good trade between almost-toll quality and low bandwith usage right? A P4 (HT) 3.2 Ghz with 1gb ram in a 800mhz board with 3 raid0 disk can sustain more than 100 calls or up to a 100? I just looking at hardware capacity, since the machine will be located at an ISP with more than needed bandwith. There is no need for voicemail, web interfaces or anything else, since the * box will only function as a gateway to a US-based VOIP provider. The machine in question runs Centos4 Linux (Redhat enterprise 4) and CDR logging only. Thanks, -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation sponsored by Easynews.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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: g729 to asterisk to g729 voip provider
On Fri, 16 Sep 2005 16:09:37 -0500 Erick Perez [EMAIL PROTECTED] wrote: Hi, your project is indeed interesting, however for learning purposes i do need to know the answer of at least: it is not my project. 1- Using sipura sip/g729 to connect to an asterisk server that will server as a gateway to a VOIP provider(g729), all in g729 will require to purchase codecs from Digium? read the page, it provides you with a free version of the g729 2- also, in this scenario the transcoding is almost non-existent right? since all codecs are the same, nothing has to be transcoded of course. 3- I have read many documents about the type of codecs, and g729 seems to be a good trade between almost-toll quality and low bandwith usage right? yes, it offers a good trade between both. however, voice quality is not superb but satisfying. regards christian On 9/16/05, ChB [EMAIL PROTECTED] wrote: Hello Erik! check out this website: http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ i have both codecs running on gentoo without problems(working with cisco 7960 and snom 190). regarding your hardware question i don't have enough experience yet, sorry. regards christian On Fri, 16 Sep 2005 15:01:32 -0500 Erick Perez [EMAIL PROTECTED] wrote: anyone with some info on this? thanks again. On 9/14/05, Erick Perez [EMAIL PROTECTED] wrote: Using sipura sip/g729 to connect to an asterisk server that will server as a gateway to a VOIP provider, all in g729 will require to purchase codecs from Digium? also, in this scenario the transcoding is almost non-existent right? I have read many documents about the type of codecs, and g729 seems to be a good trade between almost-toll quality and low bandwith usage right? A P4 (HT) 3.2 Ghz with 1gb ram in a 800mhz board with 3 raid0 disk can sustain more than 100 calls or up to a 100? I just looking at hardware capacity, since the machine will be located at an ISP with more than needed bandwith. There is no need for voicemail, web interfaces or anything else, since the * box will only function as a gateway to a US-based VOIP provider. The machine in question runs Centos4 Linux (Redhat enterprise 4) and CDR logging only. Thanks, -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation sponsored by Easynews.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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation sponsored by Easynews.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] How to access * thru router when ip address is not known
Colin Anderson Wrote: Assuming your Asterisk mail client is set up correctly, cron this twice a day on your Asterisk box: ifconfig eth0 | grep inet addr | mail -s My Asterisk IP address [EMAIL PROTECTED] Check your mail before you call and verify that the IP address has not changed, if it has, modify your client accordingly. %- SNIP! -% A more elegant method, which also works behind a firewall (if using DDNS) would be the solution discussed in the 'NAT and SIP.conf update' thread, whereby Wilson Pickett wrote: What I did was to have the ip checker write a one line file called externip.conf containing the line: externip = nnn.nnn.nnn.nnn ; this is the new ip address then in sip.conf, #include externip.conf ; replace the externip= line in sip.conf with this include Now in your ip checker script (run every n minutes with crontab),check the ip against an the last saved ip (in file old_ip for example) if ips are the same, exit, nothing to do if different, write the externip.conf file and do a sip reload command save the ip to the old_ip file Exit I have requested he send me/post his checker, as I am sure there are plenty who would find this useful and if like me do not posses the necessary skills to craft this myself. ___ --Bandwidth and Colocation sponsored by Easynews.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] Orinoco Injectors
Has anyone gotten the Lucent / Orinoco injectors (AE-1, AE-6, AE-12) to work with the Cisco 79* series phones? I'm not sure if the are the statndard POE or not -Darren ___ --Bandwidth and Colocation sponsored by Easynews.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] linux sip or iax phone that will autoanswer and route to console
Is there a linux sip or iax phone that will autoanswer and connect to the console or soundcard? I found linphonec but it does not autoanswer from what I can tell. Jerry ___ --Bandwidth and Colocation sponsored by Easynews.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] mpg123 on x86_64 (Opteron MP)
Why do you need to compile it? Isn't it available as an rpm package? -- #Joseph On Fri, 2005-09-16 at 08:56 -0700, [EMAIL PROTECTED] wrote: Has anyone successfully compiled mpg123 in the 1.0.x or 1.2beta1 distributions (I'm running FC3 linux on an Opteron 2 processor system)? Are there any patches out there to make it work? gcc -DI386_ASSEM -DPENTIUM_OPT -DREAL_IS_FLOAT -DLINUX - DREAD_MMAP -DOSS -DTERM_CONTROL-Wall -O2 - m486 -fomit-frame-pointer -funroll-all- loops -finline-functions -ffast-math -c -o dct64_i386.o dct64_i386.c as -o decode_i586.o decode_i586.s decode_i586.s: Assembler messages: decode_i586.s:44: Error: suffix or operands invalid for `push' snip decode_i586.s:161: Error: suffix or operands invalid for `pop' snip ___ --Bandwidth and Colocation sponsored by Easynews.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] Seperate Incoming calls on TDM02?
Hi Hatton, Could you provide some examples of the config files for this. Im trying to do the same. Im confused with some of the other posts (its not hard to confuse me!) Some say its just the zapata and some say theres way more to it. I have 4 FXO ports, 2 on one number and 2 on another and want to have different incoming rules\IVR depending upon channel called. Is it as simple as changing the contexts in the zapata.conf or is there more to it. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Hatton Humphrey Sent: 16 September 2005 15:17 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Seperate Incoming calls on TDM02? Yeah, in your zapata.conf just give each channel a different context setting. It's slightly more complicated with [EMAIL PROTECTED] and/or AMP, you need to use the zapata_custom.conf file, instead. You also need to use the extensions_custom.conf file, too, though there might be a better way I don't know about. I'm fighting with this right now and I'm hitting a serious frustration point - right now all incoming calls are getting handled by the from_pstn context which is how it honestly should be according to the current conf files. However when I change the context from from_pstn to aa_1 and aa_2 respectively it doesn't change anything in the way the system is answering the lines. Hatton zapata.conf: ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=no threewaycalling=no transfer=no cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include AMP configs #include zapata_additional.conf ;Include genzaptelconf configs #include zapata-auto.conf zapata_additional.conf is empty zapata-auto.conf: ; Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit ; Zaptel Channels Configurations (zapata.conf) ; ; This is not intended to be a complete zapata.conf. Rather, it is intended ; to be #include-d by /etc/zapata.conf that will include the global settings ; callerid=asreceived ; Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1 ; channel 1, WCTDM, inactive. ; channel 2, WCTDM, inactive. signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 3 context=from_pstn group=0 channel = 3 signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4 context=from_pstn group=0 channel = 4 ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] How to access * thru router when ip address is not known
Another method comes to mind similar to my last posted method, but for *behind* a firewall if you have a static hostname but dynamic ip (lots of ISP's do this): traceroute -m1 my.statichostname.net | mail -s My Asterisk IP address [EMAIL PROTECTED] Yet another method is, a lot of Linksys / SMC bluebox routers also display the external IP in their little management web site that they have. You could use this in a shell script as follows: getmyexternalip.sh rm -f /etc/myexternalip.txt wget --cache=off --http-user=admin --http-passwd=admin --output-document=/etc/myexternalip.txt http://192.168.1.1/status.html grep WAN IP Address /etc/myexternalip.txt | mail -s My Asterisk IP address [EMAIL PROTECTED] Otherwise, DyDNS.org is the way to go. ___ --Bandwidth and Colocation sponsored by Easynews.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] How to access * thru router when ip address is not known
I came up with a solution a few years ago that only required a web browser to get the current IP address. It requires an account with ssh access allowed on a web host. 1) Install your public ssh key on the account where the web pages live. 2) If you are concerned about others getting your IP address create a password protected web directory where dynamic IP host data will be stored. 3) create a script the is executed on the appropriate ip up condition. On debian this script would be installed in /etc/ppp/ip-up.d directory if you wanted it to run whenever a ppp link comes up. The script takes the output of ifconfig and uses scp to update the web host. Just browse and refresh to get the current IP address. This provides the new address right away. No need to wait for dyndns or email propagation. I should mention that I also have cron run the script every 20 minutes. I did this after I had a situation where the dsl at the web host was down for an hour or so. During that time 2 remotes changed address but were unable to update my web host. Now I will get the new address within 20 minutes of recovering from a network outage. Colin Anderson wrote: Assuming your Asterisk mail client is set up correctly, cron this twice a day on your Asterisk box: ifconfig eth0 | grep inet addr | mail -s My Asterisk IP address [EMAIL PROTECTED] Check your mail before you call and verify that the IP address has not changed, if it has, modify your client accordingly. -Original Message- From: Robert Rawlinson [mailto:[EMAIL PROTECTED] Sent: Friday, September 16, 2005 1:52 PM To: Asterisk Subject: [Asterisk-Users] How to access * thru router when ip address is not known I am going to be traveling and I wanted to be able to get on the internet and call thru * to make calls. The problem is I do not have a fixed ip address. How do you make this work? I will be using IAXCOMM. TIA Bob ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] mpg123 on x86_64 (Opteron MP)
Joseph wrote: Why do you need to compile it? Isn't it available as an rpm package? I will assume he knows why he needs to compile it. See if the source for the rpm, deb, or whatever from the distro you are running will build for you. That will often get your system to the point where any header, tools and libraries needed are now installed. That's how I do it when I want to use something from cvs. ___ --Bandwidth and Colocation sponsored by Easynews.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] TDM400P Dialing Out - Cannot be completed as dialed
I've tried to google this issue with no resolution. I'm having the same issue as this person: http://lists.digium.com/pipermail/asterisk-users/2004-August/058280.html Basically, anytime I try to dial out on my TDM400P w/ FXO, I get we're sorry, but your call cannot be completed as dialed. When I debug channel Zap/x-x, I get a whole bunch of this: [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/2-1] I recieve calls just fine. I've tried dialing the Zap group and the specific channel, all with the same results. Any ideas? zapata and zaptel configs follow. zapata.conf: [channels] context=local-in relaxdtmf=yes language=en signalling=fxs_ks ;rxwink=300 usecallerid=yes useincomingcalleridonzaptransfer=yes callerid=asreceived group=1 immediate=yes echocancel=64 echocancelwhenbridged=no echotraining=800 rxgain=9.0 txgain=1.0 channel = 1-4 zaptel.conf: fxsks=1-4 defaultzone=us loadzone=us Regards, Barry King King Computer Solutions ___ --Bandwidth and Colocation sponsored by Easynews.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] How to IGNORE distinctive ring
The way I accomplished this is to leave my default context empty, then define the distinctive ring in Asterisk, but send it to the empty default context. Asterisk will generate a warning that it doesn't know what to do, but it will also do nothing with the call.You can just do this with an empty non-default context , because * will fall back to default when it hits a completely empty context. You will need to ensure that your real incoming line go to a different context, but otherwise it should be pretty straightforward. (I don't have the configuration that I used handy, but can dig for it if my description isn't making sense.) Steve. Brad Jacobs wrote: PSI System Admin-Message-ID: [EMAIL PROTECTED] I'll give that a try in the meantime. If you find the right way to do it let me know or if I find something I'll get back to you! Thanks Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shawn Porter Sent: Tuesday, September 13, 2005 19:02 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How to IGNORE distinctive ring Brad, I posted a similar question on voipuser, no response yet, but I ended up making a separate extension Its not perfect, but it does technically ignore the call. [Home] exten = s,1,Wait(30) exten = s, 2, Hangup Shawn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brad Jacobs Sent: Tuesday, September 13, 2005 5:53 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to IGNORE distinctive ring PSI System Admin-Message-ID: [EMAIL PROTECTED] Hi list members, I'm sure this question has been posted before but I am still unable to find the answer. I have a TDM 400P line card and I would like to set it up to IGNORE the distinctive ring pattern that I have for a fax machine. Many thanks Brad -- Steven Premeau [EMAIL PROTECTED] - Gebt mir endlich einen Computer, der das tut, was ich will (und nicht was ich sage...)! ___ --Bandwidth and Colocation sponsored by Easynews.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] Grandstream
Where do I find or what is the default password for a GrandStream BT 101 for the web interface ___ --Bandwidth and Colocation sponsored by Easynews.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] Grandstream
admin? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Joshua Abbott Sent: zaterdag 17 september 2005 1:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Grandstream Where do I find or what is the default password for a GrandStream BT 101 for the web interface ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: g729 to asterisk to g729 voip provider
- Original Message - From: Christian B [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, September 17, 2005 5:44 AM Subject: Re: [Asterisk-Users] Re: g729 to asterisk to g729 voip provider On Fri, 16 Sep 2005 16:09:37 -0500 Erick Perez [EMAIL PROTECTED] wrote: Hi, your project is indeed interesting, however for learning purposes i do need to know the answer of at least: it is not my project. 1- Using sipura sip/g729 to connect to an asterisk server that will server as a gateway to a VOIP provider(g729), all in g729 will require to purchase codecs from Digium? read the page, it provides you with a free version of the g729 Erick's question is better answered here: http://www.voip-info.org/wiki-Asterisk+G.729+pass-thru Enzo ___ --Bandwidth and Colocation sponsored by Easynews.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] Double Ring
Hi, It seems like my ATA is making a ringing noise... (as it used to), but now (After the upgrade from 1.0.7 to 1.2) asterisk also is either making the ringing, or passing the PRI ringing from the telco on to me. Any suggestions on how to fix this? ___ --Bandwidth and Colocation sponsored by Easynews.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
Fwd: [Asterisk-Users] Seperate Incoming calls on TDM02?
I have 4 FXO ports, 2 on one number and 2 on another and want to have different incoming rules\IVR depending upon channel called. Is it as simple as changing the contexts in the zapata.conf or is there more to it. Here is what my experience was. Understand when reading it that I am running [EMAIL PROTECTED] version 1.5. I have 6 VoIP DIDs, two point to one auto attendant (aa_1 in my dialplan as created by AAH), two point to a second IVR (aa_2 in my dialplan as created by AAH) and the last two point to a direct extension. The From PSTN rules point all incoming calls to aa_1. AAH includes a context= in the main zapata.conf file. The rules for zapata are odd, basically you define a set of parameters and load it into a channel. If you don't redefine them and create another channel the parameters already set are included in the second channel. Therefore you can set the main setting parameters once and then change things as needed. The other thing that gets created by AAH is a file called zapata-auto.conf; this file is created by the genzaptel script and automatically sets your Zaptel channels up. If you open this script you'll see each of your incoming ports with a parameter setting of context=from_pstn. What I did (against the suggestion of the file) was change the context in the zapata-auto.conf file to point to the correct IVR. The point that I missed originally that I figured out was that the reload command in asterisk does NOT redefine the channels. You have to execute a restart command, either restart now or restart when convenient. That will reread the zapata.conf file and redefine the zaptel channels. Hope this helps, Hatton ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: Double Ring
It almost seems like I'm getting an inbound ring from my PRI/IAX terminator and asterisk is also generating a ring. If I put an 'r' in my dial statement I get only one ring. But is there any issue to be taken with putting an r in? And apparently I have to put a 'timeout' value in? On 9/16/05, Matt [EMAIL PROTECTED] wrote: Hi, It seems like my ATA is making a ringing noise... (as it used to), but now (After the upgrade from 1.0.7 to 1.2) asterisk also is either making the ringing, or passing the PRI ringing from the telco on to me. Any suggestions on how to fix this? ___ --Bandwidth and Colocation sponsored by Easynews.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] free IAX calling platform
Hello all, I have set up a free IAX calling platform similar to FreeWorldDialup/IAXtel. You can access it at http://www.goiax.com/ The website is still very beta but it will allow you to sign up for a virtual phone number, and you can make outgoing calls to US toll-free numbers. There is also a conference bridge set up. Codecs are G.711 and GSM. I intend to allow users to choose united states DIDs for free in the future after I get the website polished up a little bit. The backend is using Asterisk Realtime. If anybody has any ideas for improvements let me know. yours, Matthew TxLink ___ --Bandwidth and Colocation sponsored by Easynews.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
