RE: [Asterisk-Users] 3Com Phones
I would not recommend the 3Com phones for use with Asterisk. 3Com 3100 series phones do not support SIP with non-3Com systems. They have a basic boot loader which must download code from a 3Com NBX or a 3Com VCX system. If you don't have either of these, then you won't get runtime code on the phone, thereby making it impossible to use the thing with Asterisk. I've heard rumors that the 3103 phones have enough storage space on the phone to store a SIP image, but I don't have any more information than that. As far as 3Com licensing is concerned, it's not per year, it's per-seat (one-time charge), just like any other commercial VoIP PBX vendor (Cisco, Avaya, Shoretel, etc.) Jared Valentine [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Curt Shaffer Sent: Saturday, March 25, 2006 5:49 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] 3Com Phones I would not recommend the 3Com phones. I know to get most of them to even work on 3Com systems you need to purchase licenses. For the prices you want to pay you would definitely be better off going with something else. The list price for the 3101 is $155 The list price for the 3102 is $240 The list price for the 3103 is $365 The list price for the 3105 is $255 Phone licensing is list price of about $135/year Of course a partner could probably give you a little better of a deal depending on your relationship with them. I am freshly out of a 3Com only world so I cannot point you in the exact direction but I am sure you can get comparable phones from places like Polycom and others. Maybe these prices can give others on the list an idea of what you are looking at spending. I would stay away from anything 3Com if you want a compatible, fully functional system (Pretty scary statement from being certified in 3Com IP telephony ;)) Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of stoffell Sent: Saturday, March 25, 2006 5:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 3Com Phones On 3/25/06, Daniel Hazelbaker [EMAIL PROTECTED] wrote: We are looking at installing a VoIP system with Asterisk and are currently looking at the line of 3Com phones. Has anybody had success with using the following phones? We need to buy a lot and we don't want to end up with phones that don't work properly with asterisk. I didn't even know 3Com had VoIP phones, I'm also curious on these.. How many phones do you need and what is your budget and features wishlist? cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] free tollfree termination
Hi there, Thanks for the tip ! I am happily using this service now. One question though : I cannot get DTMF to work. Is there anything I can do in my asterisk setup to fix this ? Thanks, Lukas trixter aka Bret McDanel wrote: http://www.trxtel.com/index.php?page=Tollfree_Termination This is a free service, I am not selling anything with this service. I just thought that individuals that read this list may enjoy getting tollfree access free this way (yet another way) given that it lets you send your caller id and some of the other free providers dont let you do that. Starting a test service now, for individuals free north american tollfree termination. For carriers that do large quantities of minutes (a not really defined term as yet, more a negotiated value) we will share revenue with you for sending calls to us. If you set up IP PBX systems for customers, add a route in and make residuals off those customers. Run a ITSP? Get revenue for each minute that a customer dials a north american toll free. If anyone has any problems using the service I would appreciate hearing about it, the service will remain free even after the test period, however to get compensation requires an account so that it can be uniquely tracked. Granted tollfree traffic isnt usually the bulk of a provider, but at least now you can provide it free to your customers without losing on costs like bandwidth :) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[3]: [Asterisk-Users] Disable timeout for answered queue calls?
I'm looking for some cli output at the very moment a call on hold got
dumped.
[]'s
MM
-Original Message-
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc:
Sent: Sat, 25 Mar 2006 22:55:28 -0800 (PST)
Delivered: Sun, 26 Mar 2006 00:57:48
Subject:[Asterisk-Users] Disable timeout for answered queue calls?
queues.conf:
; How long do we let the phone ring before we consider this a timeout...
;
;timeout = 15
;
; How long do we wait before trying all the members again?
;
;retry = 5
[sales]
strategy = ringall
timeout = 300
retry = 10
member = SIP/1030
member = SIP/4000
member = SIP/4010
member = SIP/4011
extensions.conf:
[receptionist]
exten = s,1,Wait(1)
exten = s,n,NVBackgroundDetect(top-menu|t)
exten = s,n,WaitExten(15)
exten = s,n,NVBackgroundDetect(top-menu|t)
exten = s,n,WaitExten(15)
exten = s,n,Hangup
exten = 1,1,Macro(queue,Sales,sales)
exten = 2,1,Macro(queue,Tech Support,tech)
exten = 3,1,Macro(queue,Service,service)
exten = 4,1,Macro(queue,Repair,repair)
exten = 5,1,Macro(queue,Other,other)
exten = i,1,Goto(s,1)
exten = fax,1,Dial(SIP/FXS5,15)
[macro-queue]
exten = s,1,Set(CALLERID(name)=${ARG1})
exten = s,n,Set(CALLERID(number)=Hold Queue)
exten = s,n,SetMusicOnHold(moh-${ARG2})
exten = s,n,Queue(${ARG2},w)
please state exactly what cli output you are looking for.
-Dan
On Sun, 26 Mar 2006, Melcon Moraes wrote:
Hi Dan,
Paste some of your queues.conf and extensions.conf regarding to queue
and also some CLI output.
[]'s
MM
-Original Message-
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc:
Sent: Sat, 25 Mar 2006 19:32:46 -0800 (PST)
Delivered: Sat, 25 Mar 2006 21:35:08
Subject:[Asterisk-Users] Disable timeout for answered queue calls?
On Sat, 25 Mar 2006, amer karim wrote:
Hi;
Loock for ur rtpholdtimeout and rtptimeout in sip.conf.
Global Signalling Settings:
---
Codecs: none
Relax DTMF: No
Compact SIP headers:No
RTP Timeout:0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
Any other ideas?
-Dan
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Re: [Asterisk-Users] compiling Zaptel-1.2.4 on CentOS 4.3
ok got it tnx guys!On 3/26/06, Dovid Bender [EMAIL PROTECTED] wrote: Yes,There are issues witht he latest kernal release. Search the list archives. Dovid Mark Quitoriano [EMAIL PROTECTED] wrote: Hi Guys,Im having a problem compiling zaptel 1.2.4 on CentOS 4.3, anyone encountered this problem before?Here's the error i got:make -C /lib/modules/2.6.9-34.EL/build SUBDIRS=/usr/src/zaptel-1.2.4 XPPMOD= modules make[1]: Entering directory `/usr/src/kernels/2.6.9- 34.EL-i686' CC [M] /usr/src/zaptel-1.2.4/zaptel.o/usr/src/zaptel-1.2.4/zaptel.c:384: error: syntax error before zone_lock/usr/src/zaptel-1.2.4 /zaptel.c:384: warning: type defaults to `int' in declaration of `zone_lock' /usr/src/zaptel-1.2.4/zaptel.c:384: error: incompatible types in initialization/usr/src/zaptel-1.2.4/zaptel.c:384: error: initializer element is not constant /usr/src/zaptel-1.2.4/zaptel.c:384: warning: data definition has no type or storage class/usr/src/zaptel-1.2.4/zaptel.c:385: error: syntax error before chan_lock /usr/src/zaptel-1.2.4/zaptel.c:385: warning: type defaults to `int' in declaration of `chan_lock' /usr/src/zaptel-1.2.4/zaptel.c:385: error: incompatible types in initialization/usr/src/zaptel-1.2.4/zaptel.c:385: error: initializer element is not constant /usr/src/zaptel-1.2.4/zaptel.c:385: warning: data definition has no type or storage class /usr/src/zaptel-1.2.4/zaptel.c:188: warning: 'fcstab' defined but not usedmake[2]: *** [/usr/src/zaptel-1.2.4/zaptel.o] Error 1 make[1]: *** [_module_/usr/src/zaptel-1.2.4] Error 2make[1]: Leaving directory `/usr/src/kernels/2.6.9- 34.EL-i686'make: *** [linux26] Error 2-- Regards,Mark Quitoriano, CCNA Fan the flame...http://www.spreadfirefox.com/?q=user/registerr=19441 ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards,Mark Quitoriano, CCNAFan the flame... http://www.spreadfirefox.com/?q=user/registerr=19441 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 codec problems
What sort of call path are you trying to get working? Paul Hales Technical Manager AsteriskIT - Original Message - From: Rudolf Ladyzhenskii [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, March 26, 2006 10:18 AM Subject: [Asterisk-Users] G729 codec problems Hi, all I have a license for G.729A codec from Digium. When asterisk starts it shows: Jun 17 21:13:59 NOTICE[4040]: codec_g729.c:460 load_module: G.729 transcoding module Copyright (C) 1999-2005 Digium, Inc. Jun 17 21:13:59 NOTICE[4040]: codec_g729.c:461 load_module: This module is supplied under a commercial license granted by Digium, Inc. Jun 17 21:13:59 NOTICE[4040]: codec_g729.c:462 load_module: Please see the full license text supplied by the accompanying Jun 17 21:13:59 NOTICE[4040]: codec_g729.c:463 load_module: register utility, or ask for a copy from Digium. == G.729 Host-ID: cc:20:a3:86:01:93:53:92:2c:37:ae:e7:ad:16:6e:f0:39:f6:88:4e == Found license 'G729-190B962C' providing 1 channels == Found total of 1 G.729 licenses == Registered translator 'g729tolin' from format g729 to slin, cost 20 == Registered translator 'lintog729' from format slin to g729, cost 115 All is fine, however when trying to make a call I am getting: WARNING[4063]: codec_g729.c:170 g729tolin_framein: Out of G.729 Decoder Licenses! No other calls are active. Any ideas what is going on? Thanks, Rudolf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What codec extensions using now?
Hello list, Another newbie question,. if I put disallow=all and allow=g723 my sip.cof does it mean that extension could only communicate using g723 ? bellow is one of my extension example [10112] username=10112 type=friend secret=x record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=yes host=dynamic dtmfmode=rfc2833 disallow=all context=Office-lan canreinvite=no allow=g723 thanks Salaque ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hopefully a Simple Question?
Hi Guys, I'm writing an app that receives a call on an incoming channel (A), the caller negotiates through a series of prompts and is transferred to an outgoing channel (B) using the Dial cmd. That part works perfectly! For billing I'd like to be able to charge for the time that the first caller is connected to the callee on channel (B) so I can pass on my own outgoing voip costs. How do I do this? I can get the DIALTIME and END time of the call from the cdr but there doesn't seem to be a way of capturing the ANSWERTIME of channel (B) from the dialplan. Any suggestions would be greatly appreciated. clint_in_sydney ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: IAX Incoming/Outgoing
On 25 Mar 2006, at 19:15, Douglas Garstang wrote: Why do I need a username at all if I am doing rsa authentication? Why doesn't it match against the key? So you want the receiving asterisk to take an incoming key and speculatively see if it matches _any_ of the keys mentioned in it's iax.conf? Not only is that a bit expensive computationally, but it also allows an attacker to test 10 (say) keys for the price of one. Keys are for authentication not identification. Tim. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: IAX Incoming/Outgoing
Maybe you are better off with dundi ? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk spanDSP / Faxing problem
You should put something between answer and dial, to let * have time to
get the fax tone: with your dialplan, the call is immediatly bridged to
SIP/3000, so no fax detection happen at all.
A thing like:
exten = s,1,Answer
exten = s,2,Playtones(ring)
exten = s,3,Wait(3) ; if fax tone comes here, * should jump to fax extension
exten = s,4,StopPlaytones
exten = s,5,Dial(SIP/300
exten = fax,1,..
.
Hote This helps
2006/3/25, Thys de Wet [EMAIL PROTECTED]:
Hi There.I have the following setup :Asterisk 1.2.4 , freePBX 2.0.1, spandsp-0.0.2pre24My problem is as follows :If I set up a very simple extensions.conf. when I dial from a faxmachine, it seems as if no fax is being recognised.
If I answer the call, I can hear the fax machine beeping.extensions.conf :---[from-pstn]
; Answer the line and listen exten = s,1,Answer ; Dial an extension, let asterisk give a ringtone exten = s,2,Dial(SIP/3000,40,r) ; Hangup if nobody picked up within 40 seconds exten = s,3,Hangup
; Did we get a fax? --Seems as if we neverget this far :( exten = fax,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif) exten = fax,2,rxfax(${FAXFILE}|debug)
-zapata.conf :-
[trunkgroups]; define any trunk groups[channels]; hardware channels; defaultusecallerid=yeshidecallerid=nocallwaiting=nothreewaycalling=yestransfer=yesechocancel=yesechotraining=yes
faxdetect=bothbusydetect=yesbusycount=4busypattern=500,500callprogress=no; define channelscontext=from-pstn ;signalling=fxs_ls ;channel=1-3 ;context=aa_1 ;signalling=fxs_ls ;
channel=4 ;--Output from asterisk log :Mar 25 13:45:16 VERBOSE[17282]
logger.c: -- Starting simple switchon 'Zap/1-1'Mar 25 13:45:18 NOTICE[17282] chan_zap.c: Got event 18 (Ring Begin)...Mar 25 13:45:19 NOTICE[17282] chan_zap.c: Got event 2 (Ring/Answered)...Mar 25 13:45:21 NOTICE[17282] chan_zap.c: Got event 18 (Ring Begin)...
Mar 25 13:45:21 VERBOSE[17282] logger.c: -- ExecutingAnswer(Zap/1-1, ) in new stackMar 25 13:45:21 DEBUG[17282] chan_zap.c: Took Zap/1-1 off hookMar 25 13:45:21 DEBUG[17282] chan_zap.c: Enabled echo cancellation on
channel 1Mar 25 13:45:21 DEBUG[17282] chan_zap.c: Engaged echo training on channel 1Mar 25 13:45:21 VERBOSE[17282] logger.c: -- ExecutingDial(Zap/1-1, SIP/3000|40|r) in new stack
Mar 25 13:45:21 DEBUG[17282] chan_sip.c: Setting NAT on RTP to 0Mar 25 13:45:21 DEBUG[17282] chan_sip.c: Outgoing Call for 3000Mar 25 13:45:21 VERBOSE[17282] logger.c: -- Called 3000Mar 25 13:45:21 DEBUG[17282] chan_zap.c: Requested indication 3 on
channel Zap/1-1Mar 25 13:45:21 DEBUG[17229] chan_sip.c: (Provisional) Stoppingretransmission (but retaining packet) on'[EMAIL PROTECTED]
' Request 102: FoundMar 25 13:45:21 DEBUG[17229] chan_sip.c: (Provisional) Stoppingretransmission (but retaining packet) on'[EMAIL PROTECTED]
' Request 102: FoundMar 25 13:45:21 VERBOSE[17282] logger.c: -- SIP/3000-dbc4 is ringingMar 25 13:45:21 DEBUG[17222] channel.c: Avoiding initial deadlock for'SIP/3000-dbc4'Mar 25 13:45:29 DEBUG[17229] chan_sip.c: Acked pending invite 102
Mar 25 13:45:29 DEBUG[17229] chan_sip.c: Stopping retransmission on'[EMAIL PROTECTED]' of Request 102: Match Found
Mar 25 13:45:29 DEBUG[17229] chan_sip.c: build_route: Contact hop:sip:[EMAIL PROTECTED]Mar 25 13:45:29 VERBOSE[17282] logger.c: -- SIP/3000-dbc4 answeredZap/1-1
Mar 25 13:45:29 DEBUG[17282] chan_zap.c: Requested indication -1 onchannel Zap/1-1Mar 25 13:45:50 DEBUG[17282] channel.c: Didn't get a frame from channel:SIP/3000-dbc4Mar 25 13:45:50 DEBUG[17282] channel.c
: Bridge stops bridging channelsZap/1-1 and SIP/3000-dbc4Mar 25 13:45:50 DEBUG[17282] chan_sip.c: update_call_counter(3000) -decrement call limit counterMar 25 13:45:50 DEBUG[17282] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Mar 25 13:45:50 VERBOSE[17282] logger.c: == Spawn extension(from-pstn, s, 2) exited non-zero on 'Zap/1-1'Mar 25 13:45:50 DEBUG[17282] cdr_addon_mysql.c: cdr_mysql: inserting aCDR record.Mar 25 13:45:50 DEBUG[17282] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)VALUES ('2006-03-25 13:45:21','','','s','from-pstn',
'Zap/1-1','SIP/3000-dbc4','Dial','SIP/3000|40|r',29,29,'ANSWERED',3,'','1143287116.0')Mar 25 13:45:50 DEBUG[17282] chan_zap.c: Hangup: channel: 1 index = 0,normal = 10, callwait = -1, thirdcall = -1Mar 25 13:45:50 DEBUG[17282] chan_zap.c: disabled echo cancellation on
channel 1Mar 25 13:45:50 DEBUG[17282]
Re: [Asterisk-Users] RE: IAX Incoming/Outgoing
Look, you don't have to necessarily specify a username when Dial(.). It's sufficient ti specify the username in the peer declarations: On pbx1: [pbx2] type=friend username=pbx1 ; this is user for OUTGOING connections host=w.x.y.z inkeys=pbx2 outkeys=pbx1 .context= [pbx3] type=friend username=pbx1 ; this is user for OUTGOING connections host=w.x.y.z inkeys=pbx3 outkeys=pbx1 . context= On pbx2: [pbx1] type=friend username=pbx2 ; this is user for OUTGOING connections host=w.x.y.z inkeys=pbx1 outkeys=pbx2 . context= [pbx3] type=friend username=pbx2 ; this is user for OUTGOING connections host=w.x.y.z inkeys=pbx3 outkeys=pbx2 . context= On pbx3: [pbx1] type=friend username=pbx3 ; this is user for OUTGOING connections host=w.x.y.z inkeys=pbx1 outkeys=pbx3 . context= [pbx2] type=friend username=pbx3 ; this is user for OUTGOING connections host=w.x.y.z inkeys=pbx2 outkeys=pbx3 . context= Simple: 3 boxes, 3 usernames, 3 public/private key couples. Hope this helps 2006/3/25, Douglas Garstang [EMAIL PROTECTED]: I could ask why it can't authenticate against the key, but we've already been there.So, if I have 5 asterisk systems, and I want to have a different key on each, and each system has a user and a peer section, and I have to use different usernames... oh boy... this sounds like a horrible mess. -Original Message- From: Joshua Colp [mailto:[EMAIL PROTECTED]] Sent: Saturday, March 25, 2006 12:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing It still needs to know the username so it knows what entry in iax.conf to use for that information, such as the key to use. Joshua Colp - Original Message - From: Douglas Garstang [mailto: [EMAIL PROTECTED]] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-users@lists.digium.com] Sent: Sat, 25 Mar 2006 15:15:24 -0400 Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing Why do I need a username at all if I am doing rsa authentication? Why doesn't it match against the key? -Original Message- From: Joshua Colp [mailto:[EMAIL PROTECTED]] Sent: Saturday, March 25, 2006 12:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing You do realize you're not sending along a username so it's using another method to try to discover the username you're trying to authenticate as on the server side? Apparently not. IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED] Joshua Colp - Original Message - From: Douglas Garstang [mailto: [EMAIL PROTECTED]] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-users@lists.digium.com] Sent: Sat, 25 Mar 2006 14:55:28 -0400 Subject: RE: [Asterisk-Users] RE: IAX Incoming/OutgoingWell, I just tried your approach. I broke them all up into users/peers. Nowit makes even LESS sense. The pbx1 system is connecting to the pbx2 system,and according to the iax debug, is sending a username of 'pbx3_in'. *lol* [pbx1_in]type=userauth=rsainkeys=pbx1context=global_pbx_transfer deny=0.0.0.0permit=xxx.187.142.203 [pbx1_out]type=peerauth=rsa outkey=pbx1host=pbx1.ipt.yyy.com [pbx2_in]type=userauth=rsa inkeys=pbx2context=global_pbx_transferdeny=0.0.0.0permit=xxx.187.142.204 [pbx2_out]type=peerauth=rsaoutkey=pbx1host=pbx2.ipt.yyy.com [pbx3_in]type=userauth=rsainkeys=pbx3context=global_pbx_transferdeny= 0.0.0.0permit=xxx.187.142.234 [pbx3_out]type=peerauth=rsaoutkey=pbx1host= pbx3.ipt.yyy.com Here's how I connect:exten =s-CHANUNAVAIL,1,Dial(IAX2/pbx2_out/[EMAIL PROTECTED] _pbx_transfer,25,g) and here's the IAX debug:Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 3msSCall: 1DCall: 0 [xxx.187.142.204:4569] VERSION : 2 CALLED NUMBER : 2944099 CODEC_PREFS : (ulaw|g729) CALLING NUMBER: 2944093 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: Foo LANGUAGE: en CALLED CONTEXT: global_pbx_transfer FORMAT: 4 CAPABILITY: 65535 ADSICPE : 2 DATE TIME : 2006-03-2511:54:36hestia*CLI-- Called pbx2_out/[EMAIL PROTECTED]Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3msSCall: 2DCall: 1 [xxx.187.142.204:4569]Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:AUTHREQ Timestamp: 5msSCall: 2DCall: 1 [xxx.187.142.204:4569] AUTHMETHODS : 4 CHALLENGE : 129428696 USERNAME: pbx3_in WHAT THE HELL IS THIS DOINGHERE? -Original Message- From: Brian Capouch
Re: [Asterisk-Users] Error in starting * with latest trunk
On Sat, 2006-03-25 at 09:41 +0100, Dave Cotton wrote: On Sat, 2006-03-25 at 11:52 +0330, Paradise Dove wrote: hi, i've just upgraded to latest trunk. everything compiles fine but when starting this message appears and fails to start. WARNING[3990] loader.c: module chan_zap.so error /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call Here also == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log == Parsing '/etc/asterisk/dnsmgr.conf': Found Asterisk Dynamic Loader loading preload modules: Mar 25 09:28:49 WARNING[2654]: loader.c:433 check_symbols: module dir /usr/lib/asterisk/modules == Refreshing DNS lookups. Mar 25 09:28:49 WARNING[2654]: loader.c:444 check_symbols: module chan_sip.so error /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_park_call Segmentation fault (core dumped) Asterisk SVN-trunk-r14915M All running OK now but I'm not sure about this, if done a make update and it shows:- Updating from Subversion... At revision 15019. make clean make make upgrade restarting asterisk and then I get Connected to Asterisk SVN-trunk-r14988M currently running on Sheriff (pid = 32127) can someone who understands SVN more than me explain this? -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WARNING[5171]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing
why should I? i thought in 2.6 kerneles that was not necesary when you dont have physical internfaces on the system.On 3/26/06, Jonathan Augenstine [EMAIL PROTECTED] wrote: Have you verified that ztdummy is loaded?On Sun, 2006-03-26 at 01:06 -0500, Erick Perez wrote: Hi, using asterisk 1.2.5 with mysql in a centos 4.2 (2.6 kernel)no hardware interfaces installed gives me this error. Im a bit new to this so any help will be appreciated. == Parsing '/etc/asterisk/musiconhold.conf': Found Mar 26 00:58:49 WARNING[5171]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing...Sound may be choppy. [chan_oss.so] = (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found == Registered channel type 'Console' (OSS Console Channel Driver) musiconhold.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 thanks, -- --- Erick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ---Erick PerezLinux User 376588http://counter.li.org/(Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zapata configuration parsing
Hi gang. Just put an FXS port on a Zap interface for the first time. I can't figure out which parameters in zapata.conf are global and which ones can be channel specific nested. I have mucked around with it but I can't seem to make any effect on the gain levels on a per channel basis. dring1context=pbx } dring1=0,0,0} obviously global because it sets conditions for dring2context=fax } all inbound calls dring2=387,321,0} signalling=fxs_ks } is this the lead or should channel be the lead group=1 channel=1-2 rxgain=6} can this go here to effect just chan 1-2? txgain=0} signalling=fxo_ks group=2 mailbox=500 channel=3 rxgain=0 txgain=0 mailbox= channel=4 rxgain=0 txgain=0 Thanks, dbc. -- David Cook (Canada) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk spanDSP / Faxing problem
For us it takes about 6 seconds to detect fax tones, so you should change your
dialplan to either play and audio while detecting fax tones
(NVBackgroundDetect) or Wait for at least 6-7 seconds if using a Zaptel
channel.
On Sat March 25 2006 08:56, Thys de Wet wrote:
Hi There.
I have the following setup :
Asterisk 1.2.4 , freePBX 2.0.1, spandsp-0.0.2pre24
My problem is as follows :
If I set up a very simple extensions.conf. when I dial from a fax
machine, it seems as if no fax is being recognised.
If I answer the call, I can hear the fax machine beeping.
extensions.conf :
---
[from-pstn]
; Answer the line and listen
exten = s,1,Answer
; Dial an extension, let asterisk give a ringtone
exten = s,2,Dial(SIP/3000,40,r)
; Hangup if nobody picked up within 40 seconds
exten = s,3,Hangup
; Did we get a fax? --Seems as if we never
get this far :(
exten = fax,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = fax,2,rxfax(${FAXFILE}|debug)
---
-- zapata.conf :
---
-- [trunkgroups]
; define any trunk groups
[channels]
; hardware channels
; default
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
faxdetect=both
busydetect=yes
busycount=4
busypattern=500,500
callprogress=no
; define channels
context=from-pstn ;
signalling=fxs_ls ;
channel=1-3 ;
context=aa_1 ;
signalling=fxs_ls ;
channel=4 ;
---
---
Output from asterisk log :
Mar 25 13:45:16 VERBOSE[17282] logger.c: -- Starting simple switch
on 'Zap/1-1'
Mar 25 13:45:18 NOTICE[17282] chan_zap.c: Got event 18 (Ring Begin)...
Mar 25 13:45:19 NOTICE[17282] chan_zap.c: Got event 2 (Ring/Answered)...
Mar 25 13:45:21 NOTICE[17282] chan_zap.c: Got event 18 (Ring Begin)...
Mar 25 13:45:21 VERBOSE[17282] logger.c: -- Executing
Answer(Zap/1-1, ) in new stack
Mar 25 13:45:21 DEBUG[17282] chan_zap.c: Took Zap/1-1 off hook
Mar 25 13:45:21 DEBUG[17282] chan_zap.c: Enabled echo cancellation on
channel 1
Mar 25 13:45:21 DEBUG[17282] chan_zap.c: Engaged echo training on channel 1
Mar 25 13:45:21 VERBOSE[17282] logger.c: -- Executing
Dial(Zap/1-1, SIP/3000|40|r) in new stack
Mar 25 13:45:21 DEBUG[17282] chan_sip.c: Setting NAT on RTP to 0
Mar 25 13:45:21 DEBUG[17282] chan_sip.c: Outgoing Call for 3000
Mar 25 13:45:21 VERBOSE[17282] logger.c: -- Called 3000
Mar 25 13:45:21 DEBUG[17282] chan_zap.c: Requested indication 3 on
channel Zap/1-1
Mar 25 13:45:21 DEBUG[17229] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request 102: Found
Mar 25 13:45:21 DEBUG[17229] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request 102: Found
Mar 25 13:45:21 VERBOSE[17282] logger.c: -- SIP/3000-dbc4 is ringing
Mar 25 13:45:21 DEBUG[17222] channel.c: Avoiding initial deadlock for
'SIP/3000-dbc4'
Mar 25 13:45:29 DEBUG[17229] chan_sip.c: Acked pending invite 102
Mar 25 13:45:29 DEBUG[17229] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match Found
Mar 25 13:45:29 DEBUG[17229] chan_sip.c: build_route: Contact hop:
sip:[EMAIL PROTECTED]
Mar 25 13:45:29 VERBOSE[17282] logger.c: -- SIP/3000-dbc4 answered
Zap/1-1
Mar 25 13:45:29 DEBUG[17282] chan_zap.c: Requested indication -1 on
channel Zap/1-1
Mar 25 13:45:50 DEBUG[17282] channel.c: Didn't get a frame from channel:
SIP/3000-dbc4
Mar 25 13:45:50 DEBUG[17282] channel.c: Bridge stops bridging channels
Zap/1-1 and SIP/3000-dbc4
Mar 25 13:45:50 DEBUG[17282] chan_sip.c: update_call_counter(3000) -
decrement call limit counter
Mar 25 13:45:50 DEBUG[17282] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Mar 25 13:45:50 VERBOSE[17282] logger.c: == Spawn extension
(from-pstn, s, 2) exited non-zero on 'Zap/1-1'
Mar 25 13:45:50 DEBUG[17282] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR record.
Mar 25 13:45:50 DEBUG[17282] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duratio
n,billsec,disposition,amaflags,accountcode,uniqueid) VALUES ('2006-03-25
13:45:21','','','s','from-pstn',
'Zap/1-1','SIP/3000-dbc4','Dial','SIP/3000|40|r',29,29,'ANSWERED',3,'','114
3287116.0') Mar 25 13:45:50 DEBUG[17282] chan_zap.c: Hangup: channel: 1
index = 0, normal = 10, callwait = -1, thirdcall = -1
Mar 25 13:45:50 DEBUG[17282] chan_zap.c: disabled echo cancellation on
channel 1
Mar 25
Re: [Asterisk-Users] WARNING[5171]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing
Erick Perez wrote: why should I? i thought in 2.6 kerneles that was not necesary when you dont have physical internfaces on the system. ztdummy is still required for timing if you are using applications like meetme, even under the 2.6 kernel. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Copying SIP Subscriptions
On 3/26/06, Douglas Garstang [EMAIL PROTECTED] wrote: I'm pretty sure I already know the answer to this, but... Is there a way to copy/transfer/replicate sip subscriptions from one asterisk system to another, for the purposes of HA? You coudln't even write a script to do it I don't think. You can do an 'asterisk -rx sip show subscriptions' but there'd be no way to repopulate it on a second system. Yes/No? Not presently, no. But I guess the second side of this would be, if we were to copy of the subscription, how would we make it of use on the second system? would it then have to broadcast state information about the devices it was watching on system A on to system B? -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: IAX Incoming/Outgoing
On Saturday 25 March 2006 14:15, Douglas Garstang wrote: Why do I need a username at all if I am doing rsa authentication? Why doesn't it match against the key? I agree that it's suboptimal, but the IAX2 spec (at least as I understood it) REQUIRES a [EMAIL PROTECTED] I think it's silly too, but that's currently how it is. I don't know how eager Digium is to change that at this time. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polarity reversals on a TE100P
Hey guys, I've been struggling with hangup detection on a Centrex system for a bit now, I was on site Saturday and I took my DMM with me. It would appear that this Centrex service provider uses polarity reversal at the beginning of a call and several times after hangup. I've been reading the archives for the last year and it looks like there was some work done that matches my situation. With the DMM, my readings are: IDLE: -50.5 RING: 3.0 VOICE: -6.13 HANGUP 3.0 WAIT: -5.13 HANGUP 3.0 (Occurs roughly every 10 seconds) WAIT: -5.13 HANGUP 3.0 (Occurs roughly every 10 seconds) - Asterisk times out and hangs up IDLE: -50.5 I'm using an ADIT 600 with a Tellabs Echo Canceller. Is there any logging for status on the line for monitoring? Turning on hanguponpolarityswitch=yes shows that the option is ignored. I've found in the archives that this works with fxs_ks. The error goes away when setting a line line for KS, but it still isn't working My zapata.conf below: [channels] context=incoming signalling=fxs_ls callprogress=no usecallerid=no callreturn=no echocancel=no echotraining=no echocancelwhenbridged=no rxgain=12.0 txgain=-2.5 group = 1 callerid=Outside (xxx) xxx-6139 musiconhold=tape jitterbuffers=4 hanguponpolarityswitch=yes channel = 1 -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Copying SIP Subscriptions
Thanks to SER, each of our Asterisk servers knows the address of every phone. Any asterisk system can terminate a call to any phone. So, all we would need would be for the asterisk system that terminates the call to also have a copy of the subscription, and voila... it sends a NOTIFY back to the phone (well actually it sends it back to SER cuz that's where the SUBSCRIBE came from). Actually, while the above is true, I'd really like to get rid of SER alltogether. It's just an extra moving part that can break. So.. I don't know. Unfortunately Asterisk is really lacking in this area. It'd be great if there was some way to distribute registrations and subscriptions between a cluster of servers. That would rock. Doug. -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: Sun 3/26/2006 8:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Copying SIP Subscriptions On 3/26/06, Douglas Garstang [EMAIL PROTECTED] wrote: I'm pretty sure I already know the answer to this, but... Is there a way to copy/transfer/replicate sip subscriptions from one asterisk system to another, for the purposes of HA? You coudln't even write a script to do it I don't think. You can do an 'asterisk -rx sip show subscriptions' but there'd be no way to repopulate it on a second system. Yes/No? Not presently, no. But I guess the second side of this would be, if we were to copy of the subscription, how would we make it of use on the second system? would it then have to broadcast state information about the devices it was watching on system A on to system B? -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hang up when pickup analog phone
Hello,
I have a system with two cards: a HFC-PCI ISDN and a TDM21B (2 FXO and 1
FXS), running Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1l with freePBX beta5
dialplan.
I have connected an analog phone to TDM FXS port, but when I pickup the
phone to make a call, Asterisk hangs up the call. Let me explain:
In another system, when I pickup the phone, Asterisk give me tone to dial:
---
Mar 23 10:51:02 VERBOSE[4527] logger.c: -- Starting simple switch on
'Zap/1-1'
Mar 23 10:51:03 DEBUG[4527] chan_zap.c: DTMF digit: 1 on Zap/1-1
Mar 23 10:51:04 DEBUG[4527] chan_zap.c: DTMF digit: 0 on Zap/1-1
Mar 23 10:51:05 DEBUG[4527] chan_zap.c: DTMF digit: 4 on Zap/1-1
Mar 23 10:51:05 DEBUG[4527] chan_zap.c: Enabled echo cancellation on
channel 1
...
---
But in this system, when I pickup the phone, Asterisk says:
---
Mar 24 16:17:27 DEBUG[3861] chan_zap.c: Enabled echo cancellation on
channel 5
Mar 24 16:17:27 VERBOSE[3951] logger.c: -- Executing Macro(Zap/5-1,
hangupcall) in new stack
Mar 24 16:17:27 VERBOSE[3951] logger.c: -- Executing ResetCDR(Zap/5-1,
w) in new stack
Mar 24 16:17:27 DEBUG[3951] cdr_addon_mysql.c: cdr_mysql: inserting a CDR
record.
Mar 24 16:17:27 DEBUG[3951] cdr_addon_mysql.c: cdr_mysql: SQL command as
follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dst
channel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
VALUES ('2006-03-24 16:17:27','','','s','from-internal', 'Zap/5-1
','','ResetCDR','w',0,0,'NO ANSWER',3,'')
Mar 24 16:17:27 VERBOSE[3951] logger.c: -- Executing NoCDR(Zap/5-1, )
in new stack
Mar 24 16:17:27 WARNING[3951] cdr.c: CDR on channel 'Zap/5-1' not posted
Mar 24 16:17:27 WARNING[3951] cdr.c: CDR on channel 'Zap/5-1' lacks end
Mar 24 16:17:27 VERBOSE[3951] logger.c: -- Executing Wait(Zap/5-1, 5)
in new stack
Mar 24 16:17:32 VERBOSE[3951] logger.c: -- Executing Hangup(Zap/5-1, )
in new stack
Mar 24 16:17:32 VERBOSE[3951] logger.c: == Spawn extension
(macro-hangupcall, s, 4) exited non-zero on 'Zap/5-1' in macro
'hangupcall'
Mar 24 16:17:32 VERBOSE[3951] logger.c: == Spawn extension (from-internal,
s, 1) exited non-zero on 'Zap/5-1'
Mar 24 16:17:32 VERBOSE[3951] logger.c: -- Executing Macro(Zap/5-1,
hangupcall) in new stack
Mar 24 16:17:32 VERBOSE[3951] logger.c: -- Executing ResetCDR(Zap/5-1,
w) in new stack
Mar 24 16:17:32 VERBOSE[3951] logger.c: -- Executing NoCDR(Zap/5-1, )
in new stack
Mar 24 16:17:32 VERBOSE[3951] logger.c: -- Executing Wait(Zap/5-1, 5)
in new stack
Mar 24 16:17:35 DEBUG[3951] chan_zap.c: DTMF digit: 1 on Zap/5-1
Mar 24 16:17:35 DEBUG[3951] chan_zap.c: DTMF digit: 1 on Zap/5-1
Mar 24 16:17:37 VERBOSE[3951] logger.c: -- Executing Hangup(Zap/5-1, )
in new stack
Mar 24 16:17:37 VERBOSE[3951] logger.c: == Spawn extension
(macro-hangupcall, s, 4) exited non-zero on 'Zap/5-1' in macro
'hangupcall'
Mar 24 16:17:37 VERBOSE[3951] logger.c: == Spawn extension (from-internal,
h, 1) exited non-zero on 'Zap/5-1'
Mar 24 16:17:37 DEBUG[3951] chan_zap.c: Hangup: channel: 5 index = 0,
normal = 20, callwait = -1, thirdcall = -1
Mar 24 16:17:37 DEBUG[3951] chan_zap.c: disabled echo cancellation on
channel 5
Mar 24 16:17:37 DEBUG[3951] chan_zap.c: Set option TDD MODE, value: OFF(0)
on Zap/5-1
Mar 24 16:17:37 DEBUG[3951] chan_zap.c: Updated conferencing on 5, with 0
conference users
Mar 24 16:17:37 VERBOSE[3951] logger.c: -- Hungup 'Zap/5-1'
Mar 24 16:17:41 DEBUG[3861] chan_zap.c: disabled echo cancellation on
channel 5
---
I don't know why asterisk executes macro hangupcall when I pickup the
phone... I have 10 more SIP extensions that are working fine.
My zaptel.conf
---
loadzone=es
defaultzone=es
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
fxoks=5
fxsks=6-7
---
and my zapata.conf
---
[channels]
language=es
context=default
usecallerid=yes
callerid=asreceived
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
musiconhold=default
useincomingcalleridonzaptransfer=yes
; tarjeta rdsi hfc-s
signalling=bri_cpe_ptmp
switchtype=euroisdn
language=es
pridialplan=local
prilocaldialplan=local
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
nationalprefix = 0
internationalprefix = 00
faxdetect=incoming
group=0
callgroup=1
pickupgroup=1
immediate=yes
context=from-pstn
channel = 1-2
signalling=fxo_ks
context=from-internal
callerid=asreceived
callgroup=1
pickupgroup=1
group=1
channel=5
signalling=fxs_ks
context=from-pstn
faxdetect=incoming
callerid=asreceived
group=0
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
channel=6-7
---
If you need more information, please ask me. Thanks for your help.
--
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http://www.servitux.es
Tel. 966 160 600
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[Asterisk-Users] AAH: DNID not set if caller suppresses CID?
Hi, using [EMAIL PROTECTED], with quadBri from junghanns.net I am facing a strange problem: I have set incoming routes for some extension / DID: [ext-did] include = ext-did-custom exten = 23,1,SetVar(FROM_DID=23) exten = 23,2,Goto(ext-local,23,1) exten = 57,1,SetVar(FROM_DID=57) exten = 57,2,Goto(ext-local,57,1) exten = 66,1,SetVar(FROM_DID=66) exten = 66,2,Goto(ext-local,66,1) If I call from external to my * with ext 57 eveything works as expected, as long as I don't suppress my clid. Doing this, my call will not be routed to ext 57, but instead to the 'default' extension. Asterisk full log with clid: Mar 26 18:02:21 DEBUG[3582] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Mar 26 18:02:32 VERBOSE[3572] logger.c: -- Accepting voice call from '177' to '57' on channel 0/1, span 3 Mar 26 18:02:32 DEBUG[3572] chan_zap.c: Enabled echo cancellation on channel 7 Mar 26 18:02:32 VERBOSE[7387] logger.c: -- Executing SetVar(Zap/7-1, FROM_DID=57) in new stack Mar 26 18:02:32 VERBOSE[7387] logger.c: -- Executing Goto(Zap/7-1, ext-local|57|1) in new stack [...] the same w/o CLID: Mar 26 17:27:37 VERBOSE[3570] logger.c: -- Accepting voice call from '' to 's' on channel 0/1, span 1 Mar 26 17:27:37 DEBUG[3570] chan_zap.c: Enabled echo cancellation on channel 1 Mar 26 17:27:37 DEBUG[7284] pbx.c: Expression result is '1' Mar 26 17:27:37 VERBOSE[7284] logger.c: -- Executing GotoIf(Zap/1-1, 1?from-pstn-reghours|s|1:) in new stack Mar 26 17:27:37 VERBOSE[7284] logger.c: -- Goto (from-pstn-reghours,s,1) Mar 26 17:27:37 DEBUG[7284] pbx.c: Expression result is '0' Mar 26 17:27:37 VERBOSE[7284] logger.c: -- Executing GotoIf(Zap/1-1, 0?from-pstn-reghours-nofax|s|1:2) in new stack Mar 26 17:27:37 VERBOSE[7284] logger.c: -- Goto (from-pstn-reghours,s,2) Mar 26 17:27:37 VERBOSE[7284] logger.c: -- Executing Answer(Zap/1-1, ) in new stack Mar 26 17:27:37 DEBUG[7284] chan_zap.c: Engaged echo training on channel 1 Mar 26 17:27:37 VERBOSE[7284] logger.c: -- Executing Wait(Zap/1-1, 1) in new stack Mar 26 17:27:38 VERBOSE[7284] logger.c: -- Executing SetVar(Zap/1-1, intype=EXT-23) in new stack Mar 26 17:27:38 VERBOSE[7284] logger.c: -- Executing Cut(Zap/1-1, intype=intype|-|1) in new stack Can someone give me a hint what went wrong? TIA, Rgds, Hans ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3Com Phones
Drat, because the 3Com phones looked pretty good for the price. :) Is there somewhere that has a compatibility list for Asterisk with all the phones that are known to work/not work with Asterisk; since apparently VoIP phone companies incorrectly state that they support the SIP protocol (I don't consider, we support SIP as long as it only talks to our server because we tweaked it just a bit to be supported). I am looking for a good 60 phones. We are upgrading our entire phone system (and *old* NEC PBX). We don't need anything fancy on most of the phones, just the usual mid-size business features. Speakerphone, Hold, Park, Transfer, Voicemail; and we need at least 2 attendant stations that can see all in-use phone lines. We are trying to keep the costs (relatively) down, hence using Asterisk instead of a full commercial solution. It is very disconcerting to know the providers are essentially lying about what their phones support. (3Com states their phones are SIP compatible, not 3Com's version of SIP compatibile). Thanks for the info, hopefully somebody will have some recommendations for a good phone brand that actually IS Asterisk compatible. Daniel On Mar 26, 2006, at 12:01 AM, Jared Valentine wrote: I would not recommend the 3Com phones for use with Asterisk. 3Com 3100 series phones do not support SIP with non-3Com systems. They have a basic boot loader which must download code from a 3Com NBX or a 3Com VCX system. If you don't have either of these, then you won't get runtime code on the phone, thereby making it impossible to use the thing with Asterisk. I've heard rumors that the 3103 phones have enough storage space on the phone to store a SIP image, but I don't have any more information than that. As far as 3Com licensing is concerned, it's not per year, it's per- seat (one-time charge), just like any other commercial VoIP PBX vendor (Cisco, Avaya, Shoretel, etc.) Jared Valentine [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 20, Issue 184
Hi Joseph,
With iax servers dispersed across the internet, you could still use the
below setup, it would work but it's not as secure as you would want it.
I would then have a context for each server and use the IP address deny and
permit statements.
Also, you can have 1 server with a public IP and have the other servers
behind a NAT register to the public server. There are really several ways
to accomplish this. My suggestion is to start simple. Get the servers
talking with each other, successfully pass calls between themselves then add
in the security layers. Dial plan routing would probably be the same or
close to it, you could keep the same password or make it different for each
server, just be mindful of the IAX2/iaxtrunk:[EMAIL PROTECTED]/${EXTEN}
statement in the exten = Dial command, pass the correct
username:[EMAIL PROTECTED] to the correct server and you should be fine.
JR
Hi JR
Thanks for the mail , I am trying out Asterisk
and learning it , U mentioned that if all the tree
boxes are on the same subnet , there is no need for an
IAX [ context] for each outbound/inbound sessions
between the servers
If there is a situation , in which they three asterisk
box are in different locations / with separate subnet
, How will the it be
Thanks
Joseph John
Example iax.conf all PBX's
[iaxtrunk] (my internal iax trunk)
type=friend
auth=md5
secret=1234
host=dynamic
context=incomingiax
disallow=all
allow=ulaw
trunk=yes
extensions.conf all PBX's
[incomingiax]
Include = local (or whatever contexts the incoming
iax trunks need access
to)
Now routing call between them is a whole other
topic, several ways to
accomplish this but it is all dial plan related at
this point. The only
thing we accomplished so far is allowing all 3 PBX's
trunk access to each
other over a common [context] group. This is good,
as you add PBX4, PBX5,
you just add this common [context] in iax.conf in
the new servers without
the need of updating pbx1,23.
Routing example extensions.conf
[internal]
;To reach internal extensions on pbx1 (put this in
pbx 23)
Exten =
1XXX,1,Dial(IAX2/iaxtrunk:[EMAIL PROTECTED]/${EXTEN})
;To reach internal extensions on pbx2 (put this in
pbx 13)
Exten =
2XXX,1,Dial(IAX2/iaxtrunk:[EMAIL PROTECTED]/${EXTEN})
;To reach internal extensions on pbx3 (put this in
pbx 12)
Exten =
3XXX,1,Dial(IAX2/iaxtrunk:[EMAIL PROTECTED]/${EXTEN})
You could also specify each PBX in the [globals]
context
Example
[globals]
TRUNKPBX1 = IAX2/iaxtrunk:[EMAIL PROTECTED]
TRUNKPBX2 = IAX2/iaxtrunk:[EMAIL PROTECTED]
TRUNKPBX3 = IAX2/iaxtrunk:[EMAIL PROTECTED]
So your routing extension would look like this:
Exten = 1XXX,1,Dial(${TRUNKPBX1}/${EXTEN})
Exten = 2XXX,1,Dial(${TRUNKPBX2}/${EXTEN})
Exten = 3XXX,1,Dial(${TRUNKPBX3}/${EXTEN})
Hope this helps.
JR
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Re: [Asterisk-Users] 3Com Phones
Hi Daniel, If you are not locked in to an asterisk solution, I have a friend I have done a couple of network/phone systems with. I am also looking at Asterisk but have not gotten into it that far. Rich Radcliffe Kondor Waffenamt (760) 240-4728 [EMAIL PROTECTED] [EMAIL PROTECTED] 3/26/2006 9:55:38 AM Drat, because the 3Com phones looked pretty good for the price. :) Is there somewhere that has a compatibility list for Asterisk with all the phones that are known to work/not work with Asterisk; since apparently VoIP phone companies incorrectly state that they support the SIP protocol (I don't consider, we support SIP as long as it only talks to our server because we tweaked it just a bit to be supported). I am looking for a good 60 phones. We are upgrading our entire phone system (and *old* NEC PBX). We don't need anything fancy on most of the phones, just the usual mid-size business features. Speakerphone, Hold, Park, Transfer, Voicemail; and we need at least 2 attendant stations that can see all in-use phone lines. We are trying to keep the costs (relatively) down, hence using Asterisk instead of a full commercial solution. It is very disconcerting to know the providers are essentially lying about what their phones support. (3Com states their phones are SIP compatible, not 3Com's version of SIP compatibile). Thanks for the info, hopefully somebody will have some recommendations for a good phone brand that actually IS Asterisk compatible. Daniel On Mar 26, 2006, at 12:01 AM, Jared Valentine wrote: I would not recommend the 3Com phones for use with Asterisk. 3Com 3100 series phones do not support SIP with non-3Com systems. They have a basic boot loader which must download code from a 3Com NBX or a 3Com VCX system. If you don't have either of these, then you won't get runtime code on the phone, thereby making it impossible to use the thing with Asterisk. I've heard rumors that the 3103 phones have enough storage space on the phone to store a SIP image, but I don't have any more information than that. As far as 3Com licensing is concerned, it's not per year, it's per- seat (one-time charge), just like any other commercial VoIP PBX vendor (Cisco, Avaya, Shoretel, etc.) Jared Valentine [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells)
samsung, and others, I have a list, email me to remind me to dig it out and post to the list On 3/24/06, James Harper [EMAIL PROTECTED] wrote: Now that I actually try and google for it, I can't find any dual mode GSM/DECT handsets, only pages telling me that they exist without any actual information!!! Does anyone know of any such handsets? (and even better, ones that are available in Australia) I've searched a few of the major gsm manufacturers (nokia, Panasonic, sonyericsson) but their web sites are absolutely pathetic to the point being useless (or maybe I'm just in a bad mood today :) Thanks James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Harper Sent: Friday, 24 March 2006 13:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: gsm picocells Steve, Excellent explanation. In a nutshell, it might be better to just use a phone that can automatically switch between GSM and WiFi. Of course, that's limited to handful of handsets. I haven't done any sort of research, but I've been told that GSM+DECT phones are available, and while having them seamlessly switch network types during a call probably isn't possible, they can function as a cordless handset. Can anyone confirm or deny this? James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 3Com Phones
Look at the Linksys SPA942, it's a great phone for the price. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Radcliffe Sent: Sunday, March 26, 2006 10:21 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] 3Com Phones Hi Daniel, If you are not locked in to an asterisk solution, I have a friend I have done a couple of network/phone systems with. I am also looking at Asterisk but have not gotten into it that far. Rich Radcliffe Kondor Waffenamt (760) 240-4728 [EMAIL PROTECTED] [EMAIL PROTECTED] 3/26/2006 9:55:38 AM Drat, because the 3Com phones looked pretty good for the price. :) Is there somewhere that has a compatibility list for Asterisk with all the phones that are known to work/not work with Asterisk; since apparently VoIP phone companies incorrectly state that they support the SIP protocol (I don't consider, we support SIP as long as it only talks to our server because we tweaked it just a bit to be supported). I am looking for a good 60 phones. We are upgrading our entire phone system (and *old* NEC PBX). We don't need anything fancy on most of the phones, just the usual mid-size business features. Speakerphone, Hold, Park, Transfer, Voicemail; and we need at least 2 attendant stations that can see all in-use phone lines. We are trying to keep the costs (relatively) down, hence using Asterisk instead of a full commercial solution. It is very disconcerting to know the providers are essentially lying about what their phones support. (3Com states their phones are SIP compatible, not 3Com's version of SIP compatibile). Thanks for the info, hopefully somebody will have some recommendations for a good phone brand that actually IS Asterisk compatible. Daniel On Mar 26, 2006, at 12:01 AM, Jared Valentine wrote: I would not recommend the 3Com phones for use with Asterisk. 3Com 3100 series phones do not support SIP with non-3Com systems. They have a basic boot loader which must download code from a 3Com NBX or a 3Com VCX system. If you don't have either of these, then you won't get runtime code on the phone, thereby making it impossible to use the thing with Asterisk. I've heard rumors that the 3103 phones have enough storage space on the phone to store a SIP image, but I don't have any more information than that. As far as 3Com licensing is concerned, it's not per year, it's per- seat (one-time charge), just like any other commercial VoIP PBX vendor (Cisco, Avaya, Shoretel, etc.) Jared Valentine [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax limit question
I found a solution... I just has to enter an Answer
line and now it behaves as I wanted. Here is the
working code:
[inbound]
exten = 1234567,1,Set(GROUP()=limit)
exten = 1234567,2,GotoIf($[${GROUP_COUNT()}2]?103)
exten = 1234567,3,Dial(Zap/5Zap/6,25,tT)
exten = 1234567,4,Voicemail,u110
exten = 1234567,5,hangup
exten = 1234567,103,Answer
exten = 1234567,104,Playtones(busy)
exten = 1234567,105,Wait(5)
exten = 1234567,106,Hangup
DB
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Re: [Asterisk-Users] 3Com Phones
3Com is one of the few that lie about it. Many Cisco phones support SIP, but not all of them. I think Nortel also lies about SIP on some of their phones. Daniel Hazelbaker wrote: Drat, because the 3Com phones looked pretty good for the price. :) Is there somewhere that has a compatibility list for Asterisk with all the phones that are known to work/not work with Asterisk; since apparently VoIP phone companies incorrectly state that they support the SIP protocol (I don't consider, we support SIP as long as it only talks to our server because we tweaked it just a bit to be supported). I am looking for a good 60 phones. We are upgrading our entire phone system (and *old* NEC PBX). We don't need anything fancy on most of the phones, just the usual mid-size business features. Speakerphone, Hold, Park, Transfer, Voicemail; and we need at least 2 attendant stations that can see all in-use phone lines. We are trying to keep the costs (relatively) down, hence using Asterisk instead of a full commercial solution. It is very disconcerting to know the providers are essentially lying about what their phones support. (3Com states their phones are SIP compatible, not 3Com's version of SIP compatibile). Thanks for the info, hopefully somebody will have some recommendations for a good phone brand that actually IS Asterisk compatible. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3Com Phones
If you can find yourself a local Asterisk consultant, they should be able to let you see some phones and maybe even try them out. Paul Hales Technical Manager AsteriskIT - Original Message - From: Daniel Hazelbaker [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 3:55 AM Subject: Re: [Asterisk-Users] 3Com Phones Drat, because the 3Com phones looked pretty good for the price. :) Is there somewhere that has a compatibility list for Asterisk with all the phones that are known to work/not work with Asterisk; since apparently VoIP phone companies incorrectly state that they support the SIP protocol (I don't consider, we support SIP as long as it only talks to our server because we tweaked it just a bit to be supported). I am looking for a good 60 phones. We are upgrading our entire phone system (and *old* NEC PBX). We don't need anything fancy on most of the phones, just the usual mid-size business features. Speakerphone, Hold, Park, Transfer, Voicemail; and we need at least 2 attendant stations that can see all in-use phone lines. We are trying to keep the costs (relatively) down, hence using Asterisk instead of a full commercial solution. It is very disconcerting to know the providers are essentially lying about what their phones support. (3Com states their phones are SIP compatible, not 3Com's version of SIP compatibile). Thanks for the info, hopefully somebody will have some recommendations for a good phone brand that actually IS Asterisk compatible. Daniel On Mar 26, 2006, at 12:01 AM, Jared Valentine wrote: I would not recommend the 3Com phones for use with Asterisk. 3Com 3100 series phones do not support SIP with non-3Com systems. They have a basic boot loader which must download code from a 3Com NBX or a 3Com VCX system. If you don't have either of these, then you won't get runtime code on the phone, thereby making it impossible to use the thing with Asterisk. I've heard rumors that the 3103 phones have enough storage space on the phone to store a SIP image, but I don't have any more information than that. As far as 3Com licensing is concerned, it's not per year, it's per- seat (one-time charge), just like any other commercial VoIP PBX vendor (Cisco, Avaya, Shoretel, etc.) Jared Valentine [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells)
If you find anything out, I would like to know. I have tried to find a gsm/wifi phone in the past (in melbourne) and failed. later, Paul Hales Technical Manager AsteriskIT - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 25, 2006 11:21 AM Subject: [Asterisk-Users] GSM/DECT handsets (was gsm picocells) Now that I actually try and google for it, I can't find any dual mode GSM/DECT handsets, only pages telling me that they exist without any actual information!!! Does anyone know of any such handsets? (and even better, ones that are available in Australia) I've searched a few of the major gsm manufacturers (nokia, Panasonic, sonyericsson) but their web sites are absolutely pathetic to the point being useless (or maybe I'm just in a bad mood today :) Thanks James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Harper Sent: Friday, 24 March 2006 13:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: gsm picocells Steve, Excellent explanation. In a nutshell, it might be better to just use a phone that can automatically switch between GSM and WiFi. Of course, that's limited to handful of handsets. I haven't done any sort of research, but I've been told that GSM+DECT phones are available, and while having them seamlessly switch network types during a call probably isn't possible, they can function as a cordless handset. Can anyone confirm or deny this? James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MusicOnHold with mpg123
Alright, I've come across a really strange issue and I've been banging my head trying to figure it out. I have 3 machines. 1 Dell Dimension 4100, Pentium 3. 1 Dell 400SC, Pentium 4. 1 Dell 1600SC, Xeon. I run mpg123 0.59r on each machine. Using RH9 with a 2.4.20-8 kernel, each machine plays MoH flawlessly. As RH9 gets older and older, however, the need to upgrade arose. So I upgraded each machine to CentOS 4.3, with various 2.6 kernels ranging from 2.6.22 to 2.6.34 (yes, with the spinlock error). With these machines in this current state, the old Dell Dimension plays MoH flawlessly (the provided fpm MoH with the distro), and the two newer dells (400SC and 1600SC) all have heavy static, crackling, and other undesirable noises introduced into what I can only guess is the decoding of the MP3 files. Neither of these newer machines can transcode between audio file formats without introducing this same static using sox as well. I tried moving to native MoH, using the format_mp3 module, but found the lack of volume control to be problematic for us. Since we've been dealing with either no MoH, or moh that has undesirable qualities (too loud, some static/crackling, etc) If anyone has some ideas regarding this, I'd be happy to hear them. Then maybe Jared and Leif won't have to put up with my exasperated, repeated attempts at fixing this :) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP realtime: how to authenticate without name field ?
Hi, Can someone explain to me how to set up the sip_buddies table from 1.2.5 properly so my users can authenticate correctly without using the name field ? (if it's possible) First I was assuming that it would be possible for a user to connect and dial just providing username,secret,host and context but it seems that I need name to be set as well. If I set the name field, the user can authenticate ; the username field is even updated in the database to reflect the name field, this automatically. Now I think this is giving me a problem because I loose the name information that is stored in the various ATAs, and I would like to get it back in my CDRs, so I thought of removing the info from the name field in the database since it is overriding the ATA info, and provide only the username field for authentication ; but, it doesn't work. Is it an option in the ATA that I should change so it could authenticate only with username field ? Or is it something in my config/sip_buddies table ? Thanks in advance for the attention, Frederic ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] tsu-600
i wrote previous about a setup i thought might work with asterisk and the tsu-600. no one replied, so i thought i would ask if anyone is using a tsu-600 with asterisk and if so how do you have it setup ?? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hopefully a Simple Question?
What about using system(echo) to push stuff into a text file, or the mysql plugin to push stuff over to a database? Paul Hales Technical Manager AsteriskIT - Original Message - From: Clint Tevlin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, March 26, 2006 9:36 PM Subject: [Asterisk-Users] Hopefully a Simple Question? Hi Guys, I'm writing an app that receives a call on an incoming channel (A), the caller negotiates through a series of prompts and is transferred to an outgoing channel (B) using the Dial cmd. That part works perfectly! For billing I'd like to be able to charge for the time that the first caller is connected to the callee on channel (B) so I can pass on my own outgoing voip costs. How do I do this? I can get the DIALTIME and END time of the call from the cdr but there doesn't seem to be a way of capturing the ANSWERTIME of channel (B) from the dialplan. Any suggestions would be greatly appreciated. clint_in_sydney ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help on mfc/r2
Melcon Moraes napisał(a): supertones=pl inside your unicall.conf Ok, done. something missing, isn't? What are you trying to do? Maybe that will be better (thz Marcin): # cat call Channel: Unicall/1/363 Application: playback Data: demo-thanks klaudia*CLI !cp call /var/spool/asterisk/outgoing -- Attempting call on Unicall/1/363 for application playback(demo-thanks) (Retry 1) Mar 26 23:49:32 WARNING[13415]: chan_unicall.c:634 unicall_report: MFC/R2 UniCall/1 Call control(1) Mar 26 23:49:32 WARNING[13415]: chan_unicall.c:634 unicall_report: MFC/R2 UniCall/1 Make call Mar 26 23:49:32 WARNING[13415]: chan_unicall.c:1084 unicall_call: Make call failed - Blocked Mar 26 23:49:32 NOTICE[13415]: channel.c:2435 __ast_request_and_dial: Unable to call channel Unicall/1/363 Mar 26 23:49:32 WARNING[13415]: chan_unicall.c:634 unicall_report: MFC/R2 UniCall/1 Channel gains Mar 26 23:49:32 WARNING[13415]: chan_unicall.c:634 unicall_report: MFC/R2 UniCall/1 Channel switching -- Hungup 'UniCall/1-1' Mar 26 23:49:32 NOTICE[13415]: pbx_spool.c:269 attempt_thread: Call failed to go through, reason 0 363 is my regular phone number inside company. If you give me a little more info, I promise I'll try to help you. :) I use PCM R/2 in Alcatel OXE. (On Big Alcatel 4400, aka The Crystal, aka ACT it would be called PCM2). on the OXE side PCM/R2 is configured like that: ┌─Consult/Modify: Board──┐ │ │ │ Node Number (reserved) : 102 │ │ Shelf Address : 1 │ │ Board Address : 4 │ │ │ │ Interface Type + MG-IVR Z30 │ │ Virtual board + NO │ │ Serial number : -- │ │ Usage State + Busy │ │ Operational State + Enabled │ │ Main/Standby State + Main (Master) │ │ Number Of Sets Being Connect. : 30 │ │ Country Protocol Type + Poland │ │ Send Init Dynamic Msg + False │ │ Param By Default + False │ │ Incidents Teleservice + NO │ │ IVR Protocol + OPS-FX Protocol │ │ Network recording use + False │ │ │ └┘ -- Krzysztof Drewicz [EMAIL PROTECTED] +48 22 34 54 363 begin:vcard fn:Krzysztof Drewicz n:Drewicz;Krzysztof adr;quoted-printable:;;Wybrze=C5=BCe Gdy=C5=84skie 6C;Warszawa;Mazowieckie;01-531;Polska email;internet:[EMAIL PROTECTED] tel;work:+48 22 34 54 363 tel;cell:+48 606 698 295 x-mozilla-html:TRUE url:http://www.citicom.pl version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK EI
I'm using a Digium TE411P connected to a UK switch (EuroISDN). Everything is working, but if I dial a busy number (from SIP) is seems to stay busy until I hang up, even though the dial-plan drops through some other stuff using CALLSTATUS variable (i.e. S-BUSY), none of the timeouts come into play. Any ideas? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3Com Phones
Daniel Hazelbaker wrote: Drat, because the 3Com phones looked pretty good for the price. :) Good for the price? You can import an atcom AT-320 EE for $40 +pp (although they are hardly fantastic phones, at least they support IAX2). They have a few faults (the speed-dial keys aren't really speed dial keys, they are a replication of the keypad ones, there is no headset port on any model, no backlight on the display, the telnet client is hopeless, the built-in webserver is ugly and you can't reconfigure them en masse, and it has buttons on it that although nice, I can't for the life of me work out what use they are) but they are solidly built and don't look nearly as ugly as a 3c 3101. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tsu-600
On Sun, 26 Mar 2006, mike webb wrote: i wrote previous about a setup i thought might work with asterisk and the tsu-600. no one replied, so i thought i would ask if anyone is using a tsu-600 with asterisk and if so how do you have it setup ?? The Adtran TSU-600 can be made to work like a normal channel bank. I'm not using one with Asterisk, but the Adtran docs are very well written. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Jittery Linksys/Sipura meetme conference fixed
Ina previous message I described how Linksys 942 phone users who dialed in to a meetme conference at their site heard severe jitter. This was also experienced with Sipura SPA-2002 ATAs. Users of other IP phones like Polycom and Snom had no such problem. Also, the Linksys and SPA users had no problems with regular phone calls, just the meetme conference. This problem was finally fixed by going in to the settings for the Linksys phone and setting the RTP frame length to 0.020 and disabling the jitter buffer adjustment. The same fix also worked for the SPA. Hope this helps others who have experienced a similar problem. Rana Dutt Softel Solutions www.softelinc.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 301 is slow
On Sat March 25 2006 18:06, Nick Hoffman [EMAIL PROTECTED] wrote: Hi guys, I've been using a Polycom IP 301 for a couple of weeks now and find that it's extremely slow for configuring. For instance, it takes several minutes to boot up, apply any changes via the web interface takes at least a minute, etc. Is this normal behaviour? Is there anything that can be done about it? On Sun March 26 2006 05:35, [EMAIL PROTECTED] wrote: polycoms are just that way. they are glacially slow on rebooting, and reboot for any trivial change. and no, nothing can be done about it. -Dan On Sun March 26 2006 06:20, Kevin P. Fleming [EMAIL PROTECTED] wrote: It is normal, and there is nothing you can do about it. The processor in the Polycom phones is not fast... Hrm, well that's disappointing. If they're so slow, why are they so popular? -- Nick e: [EMAIL PROTECTED] p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tsu-600
mike webb wrote: i wrote previous about a setup i thought might work with asterisk and the tsu-600. no one replied, so i thought i would ask if anyone is using a tsu-600 with asterisk and if so how do you have it setup ?? ___ I have three working. The work fine except there is no callerid on the units I got. What else do you need? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 301 is slow
Easy to configure, lots of options and features, excellent quality speaker for hands free. Although the 301 is nothing to get excited about. The 501 and 601 are much better. Doug. -Original Message- From: Nick Hoffman [mailto:[EMAIL PROTECTED] Sent: Sun 3/26/2006 4:57 PM To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED]; Kevin P. Fleming Subject: Re: [Asterisk-Users] Polycom IP 301 is slow On Sat March 25 2006 18:06, Nick Hoffman [EMAIL PROTECTED] wrote: Hi guys, I've been using a Polycom IP 301 for a couple of weeks now and find that it's extremely slow for configuring. For instance, it takes several minutes to boot up, apply any changes via the web interface takes at least a minute, etc. Is this normal behaviour? Is there anything that can be done about it? On Sun March 26 2006 05:35, [EMAIL PROTECTED] wrote: polycoms are just that way. they are glacially slow on rebooting, and reboot for any trivial change. and no, nothing can be done about it. -Dan On Sun March 26 2006 06:20, Kevin P. Fleming [EMAIL PROTECTED] wrote: It is normal, and there is nothing you can do about it. The processor in the Polycom phones is not fast... Hrm, well that's disappointing. If they're so slow, why are they so popular? -- Nick e: [EMAIL PROTECTED] p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] help on mfc/r2
Now we're talking. :) I don't know anythig about Alcatel boxes, but can you make a simple call to your regular phone number from some SIP/IAX2 softphone/hardphone? What do you have in zaptel.conf file? For instance, I have this on my: span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 And those bits after the channel range sets the state of each channel. I don't know if they are the same for your box but, blocked has something to do with that. What happens if you use Unicall/g1 instead of just channel 1 of Unicall? Do you have zttool compiled? How those bits are showed up? The timing source is correctly set up? Seems that your master timing sync source is the Alcatel, so you should use span=1,1,0 It looks like I'm filling you with more questions than answers. :) []'s MM -Original Message- From: Krzysztof Drewicz [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Mon, 27 Mar 2006 01:01:12 +0200 Delivered: Sun, 26 Mar 2006 16:59:27 Subject:[Asterisk-Users] help on mfc/r2 Melcon Moraes napisaÅ(a): supertones=pl inside your unicall.conf Ok, done. something missing, isn't? What are you trying to do? Maybe that will be better (thz Marcin): # cat call Channel: Unicall/1/363 Application: playback Data: demo-thanks klaudia*CLI !cp call /var/spool/asterisk/outgoing -- Attempting call on Unicall/1/363 for application playback(demo-thanks) (Retry 1) Mar 26 23:49:32 WARNING[13415]: chan_unicall.c:634 unicall_report: MFC/R2 UniCall/1 Call control(1) Mar 26 23:49:32 WARNING[13415]: chan_unicall.c:634 unicall_report: MFC/R2 UniCall/1 Make call Mar 26 23:49:32 WARNING[13415]: chan_unicall.c:1084 unicall_call: Make call failed - Blocked Mar 26 23:49:32 NOTICE[13415]: channel.c:2435 __ast_request_and_dial: Unable to call channel Unicall/1/363 Mar 26 23:49:32 WARNING[13415]: chan_unicall.c:634 unicall_report: MFC/R2 UniCall/1 Channel gains Mar 26 23:49:32 WARNING[13415]: chan_unicall.c:634 unicall_report: MFC/R2 UniCall/1 Channel switching -- Hungup 'UniCall/1-1' Mar 26 23:49:32 NOTICE[13415]: pbx_spool.c:269 attempt_thread: Call failed to go through, reason 0 363 is my regular phone number inside company. If you give me a little more info, I promise I'll try to help you. :) I use PCM R/2 in Alcatel OXE. (On Big Alcatel 4400, aka The Crystal, aka ACT it would be called PCM2). on the OXE side PCM/R2 is configured like that: ââConsult/Modify: Boardâââââââââââââââââââââââââââââââââââââââââââââââ â â â Node Number (reserved) : 102 â â Shelf Address : 1 â â Board Address : 4 â â â â Interface Type + MG-IVR Z30 â â Virtual board + NO â â Serial number : -- â â Usage State + Busy â â Operational State + Enabled â â Main/Standby State + Main (Master) â â Number Of Sets Being Connect. : 30 â â Country Protocol Type + Poland â â Send Init Dynamic Msg + False â â Param By Default + False â â Incidents Teleservice + NO â â IVR Protocol + OPS-FX Protocol â â Network recording use + False â â â ââââââââââââââââââââââââââââââââââââââââââââââââââââââââââââââââââââââ -- Krzysztof Drewicz [EMAIL PROTECTED] +48 22 34 54 363 E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,1143413967.899910.29065.alcuta.terra.com.br,6811,Des15,Des15 --Original Message Ends-- -- Melcon Moraes [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 301 is slow
Nick Hoffman wrote: Hrm, well that's disappointing. If they're so slow, why are they so popular? They may be slow to startup, but they're great phones. :) Once the phone has started up, it works like a charm and the sound/call quality is fantastic. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells)
Not GSM/DECT but GSM/Wifi phones are available - This is not a recommendation, I don't like what I've seen. try www.imate.com (to start with) .. they have at least three types of GSM phones that do Wifi .. They run windows so there are several sip softwares and one IAX software that work with these - Also Nokia has a GSM phone that does Wifi but that's a symbian (OS) phone (don't know of sip software that works with it). - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 00:48 Subject: Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells) If you find anything out, I would like to know. I have tried to find a gsm/wifi phone in the past (in melbourne) and failed. later, Paul Hales Technical Manager AsteriskIT - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 25, 2006 11:21 AM Subject: [Asterisk-Users] GSM/DECT handsets (was gsm picocells) Now that I actually try and google for it, I can't find any dual mode GSM/DECT handsets, only pages telling me that they exist without any actual information!!! Does anyone know of any such handsets? (and even better, ones that are available in Australia) I've searched a few of the major gsm manufacturers (nokia, Panasonic, sonyericsson) but their web sites are absolutely pathetic to the point being useless (or maybe I'm just in a bad mood today :) Thanks James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Harper Sent: Friday, 24 March 2006 13:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: gsm picocells Steve, Excellent explanation. In a nutshell, it might be better to just use a phone that can automatically switch between GSM and WiFi. Of course, that's limited to handful of handsets. I haven't done any sort of research, but I've been told that GSM+DECT phones are available, and while having them seamlessly switch network types during a call probably isn't possible, they can function as a cordless handset. Can anyone confirm or deny this? James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells)
I think the main issue for James and myself is that we can't buy anything in Australia. Paul Hales Technical Manager AsteriskIT - Original Message - From: AR Tarzi [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 10:21 AM Subject: Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells) Not GSM/DECT but GSM/Wifi phones are available - This is not a recommendation, I don't like what I've seen. try www.imate.com (to start with) .. they have at least three types of GSM phones that do Wifi .. They run windows so there are several sip softwares and one IAX software that work with these - Also Nokia has a GSM phone that does Wifi but that's a symbian (OS) phone (don't know of sip software that works with it). - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 00:48 Subject: Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells) If you find anything out, I would like to know. I have tried to find a gsm/wifi phone in the past (in melbourne) and failed. later, Paul Hales Technical Manager AsteriskIT - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 25, 2006 11:21 AM Subject: [Asterisk-Users] GSM/DECT handsets (was gsm picocells) Now that I actually try and google for it, I can't find any dual mode GSM/DECT handsets, only pages telling me that they exist without any actual information!!! Does anyone know of any such handsets? (and even better, ones that are available in Australia) I've searched a few of the major gsm manufacturers (nokia, Panasonic, sonyericsson) but their web sites are absolutely pathetic to the point being useless (or maybe I'm just in a bad mood today :) Thanks James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Harper Sent: Friday, 24 March 2006 13:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: gsm picocells Steve, Excellent explanation. In a nutshell, it might be better to just use a phone that can automatically switch between GSM and WiFi. Of course, that's limited to handful of handsets. I haven't done any sort of research, but I've been told that GSM+DECT phones are available, and while having them seamlessly switch network types during a call probably isn't possible, they can function as a cordless handset. Can anyone confirm or deny this? James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 301 is slow
And the fact that rebooting a phone is a fairly rare occurence. Paul Hales Technical Manager AsteriskIT - Original Message - From: Avi Miller [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED]; Kevin P. Fleming [EMAIL PROTECTED] Sent: Monday, March 27, 2006 10:17 AM Subject: Re: [Asterisk-Users] Polycom IP 301 is slow Nick Hoffman wrote: Hrm, well that's disappointing. If they're so slow, why are they so popular? They may be slow to startup, but they're great phones. :) Once the phone has started up, it works like a charm and the sound/call quality is fantastic. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Hopefully a Simple Question?
Hi Guys, I'm writing an app that receives a call on an incoming channel (A), the caller negotiates through a series of prompts and is transferred to an outgoing channel (B) using the Dial cmd. That part works perfectly! For billing I'd like to be able to charge for the time that the first caller is connected to the callee on channel (B) so I can pass on my own outgoing voip costs. How do I do this? I can get the DIALTIME and END time of the call from the cdr but there doesn't seem to be a way of capturing the ANSWERTIME of channel (B) from the dialplan. Any suggestions would be greatly appreciated. clint_in_sydney * Clint, Use the forkcdr command in the extension logic right before you connect the caller to channel B. This will close the cdr entry for the incoming call (going through he prompts) and start a new cdr entry for the outgoing call. Works great for me. JR ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help on mfc/r2
Melcon Moraes napisał(a): but can you make a simple call to your regular phone number from some SIP/IAX2 softphone/hardphone? Right now? It's not working. I could use (i have needed hardware). Data: SIP/extension-number Or even capi/zap channel. When e400p is configured as a E1/PRI and connected to diffrent card, my .call-file works (I hear asterisk-demo after receivng phone). So i assume my .call file is OK. What do you have in zaptel.conf file? For instance, I have this on my: span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 Me too. I've seen dchannel=16 somewhere on the net, but it makes me very confused, as R2 imvho is beeing digigit-digit on every litle time-slot/channel. hing to do with that. What happens if you use Unicall/g1 instead of just channel 1 of Unicall? Same, it's still: *CLI -- Attempting call on Unicall/g1/363 for application playback(demo-thanks) (Retry 1) Mar 27 01:42:00 NOTICE[13840]: channel.c:2437 __ast_request_and_dial: Unable to request channel Unicall/g1/363 Mar 27 01:42:00 NOTICE[13840]: pbx_spool.c:269 attempt_thread: Call failed to go through, reason 0 Do you have zttool compiled? How those bits are showed up? The timing │ Current Alarms: No alarms. │ │ Sync Source: Tormenta 2 (PCI) Quad E1 Card │ │ IRQ Misses: 0 │ │ Bipolar Viol: 609 │ │ Tx/Rx Levels: 0/ 0 │ │ Total/Conf/Act: 31/ 30/ 30 │ │ 112333 ┌──┐ │ │ 1234567890123456789012345789012 │ Back │ │ │ TxA 111 111 └──┘ │ │ TxB 111 111 │ │ TxC 000 000 │ │ TxD 111 111 │ │ ┌──┐ │ │ RxA 000 000 │ Loop │ │ │ RxB 111 111 └──┘ │ │ RxC 000 000 │ │ RxD 111 111 │ │ │ source is correctly set up? Seems that your master timing sync source is the Alcatel, so you should use span=1,1,0 ok, -- Krzysztof Drewicz [EMAIL PROTECTED] +48 22 34 54 363 begin:vcard fn:Krzysztof Drewicz n:Drewicz;Krzysztof adr;quoted-printable:;;Wybrze=C5=BCe Gdy=C5=84skie 6C;Warszawa;Mazowieckie;01-531;Polska email;internet:[EMAIL PROTECTED] tel;work:+48 22 34 54 363 tel;cell:+48 606 698 295 x-mozilla-html:TRUE url:http://www.citicom.pl version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 360 - Multiple Server BLF Indications
Hi, This is a weird request, but does anyone have a Snom 360 monitoring extensions for BLF on several Asterisk servers accross a network? Alternatively, can anyone give me a pointer as to how to setup a Snom 360 to monitor an extension not on it's own server? My scenario is that I have a main site which will have its own server (for storage of call recording data etc because the remote sites don't have the appropriate facilities) and each site has its own embedded system (to ensure that if the network goes down we can still use a normal telephone line). We need an operator telephone with expansion modules (hence the Snom 360) to monitor approximately 180 extensions on approximately 60 asterisk systems (about three extensions per site) so the operator can immediately see any extensions that successfully initiate a call. Any information would be greatly appreciated. Kind Regards Stuart begin:vcard fn:Stuart Elvish n:Elvish;Stuart org:Dallas Delta Corporation Pty Ltd;Voice Networking Directorate adr:;;102 Albert Street;East Brunswick;VIC;3057;Australia email;internet:[EMAIL PROTECTED] title:Voice Networking Engineer tel;work:03 9387 7445 tel;fax:03 9387 3128 tel;cell:0408 873 601 x-mozilla-html:TRUE url:http://www.dallasdelta.net version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications
I have a bad feeling that getting a phone with 160 lights is not going to happen anytime soon. From memory, the snom360 is limited to way less than that. Paul Hales Technical Manager AsteriskIT - Original Message - From: Stuart Elvish - Dallas Delta Corporation Pty Ltd [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 10:57 AM Subject: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications Hi, This is a weird request, but does anyone have a Snom 360 monitoring extensions for BLF on several Asterisk servers accross a network? Alternatively, can anyone give me a pointer as to how to setup a Snom 360 to monitor an extension not on it's own server? My scenario is that I have a main site which will have its own server (for storage of call recording data etc because the remote sites don't have the appropriate facilities) and each site has its own embedded system (to ensure that if the network goes down we can still use a normal telephone line). We need an operator telephone with expansion modules (hence the Snom 360) to monitor approximately 180 extensions on approximately 60 asterisk systems (about three extensions per site) so the operator can immediately see any extensions that successfully initiate a call. Any information would be greatly appreciated. Kind Regards Stuart ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications
I installed 2 Snom360's a few months ago, and 'at the time' only 1 expansion module could be added. (also the fact that the modules draw so much current that it got the POE switch upset!) Have you tested a snom360? I should have one in the lab soon enough. Paul Hales Technical Manager AsteriskIT - Original Message - From: Stuart Elvish - Dallas Delta Corporation Pty Ltd [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 11:41 AM Subject: Re: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications There is an add on module for this phone and according to a source that distributes them here, the modules can be daisy chained until you reach the required number of extensions. I didn't think you could, but that is the information that we have at hand... [EMAIL PROTECTED] wrote: I have a bad feeling that getting a phone with 160 lights is not going to happen anytime soon. From memory, the snom360 is limited to way less than that. Paul Hales Technical Manager AsteriskIT - Original Message - From: Stuart Elvish - Dallas Delta Corporation Pty Ltd [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 10:57 AM Subject: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications Hi, This is a weird request, but does anyone have a Snom 360 monitoring extensions for BLF on several Asterisk servers accross a network? Alternatively, can anyone give me a pointer as to how to setup a Snom 360 to monitor an extension not on it's own server? My scenario is that I have a main site which will have its own server (for storage of call recording data etc because the remote sites don't have the appropriate facilities) and each site has its own embedded system (to ensure that if the network goes down we can still use a normal telephone line). We need an operator telephone with expansion modules (hence the Snom 360) to monitor approximately 180 extensions on approximately 60 asterisk systems (about three extensions per site) so the operator can immediately see any extensions that successfully initiate a call. Any information would be greatly appreciated. Kind Regards Stuart -- -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 301 is slow
The worst thing on all Polycom IP phones is the speaker phone's poor quality. You could not have a conference call using the speakers, only the head phone. Denis. On 26 de mar de 2006, at 21:17, Avi Miller wrote: Nick Hoffman wrote: Hrm, well that's disappointing. If they're so slow, why are they so popular? They may be slow to startup, but they're great phones. :) Once the phone has started up, it works like a charm and the sound/call quality is fantastic. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications
I had a look at the snom website - and the manual for the expansion module read that only one module can be attached 'currently'. So maybe this has changed. Any ideas? Personally, I like snom phones a lot. I used a snom 200 at my desk at a previous job for almost 2 years. Paul Hales Technical Manager AsteriskIT - Original Message - From: Stuart Elvish - Dallas Delta Corporation Pty Ltd [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 11:41 AM Subject: Re: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications There is an add on module for this phone and according to a source that distributes them here, the modules can be daisy chained until you reach the required number of extensions. I didn't think you could, but that is the information that we have at hand... [EMAIL PROTECTED] wrote: I have a bad feeling that getting a phone with 160 lights is not going to happen anytime soon. From memory, the snom360 is limited to way less than that. Paul Hales Technical Manager AsteriskIT - Original Message - From: Stuart Elvish - Dallas Delta Corporation Pty Ltd [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 10:57 AM Subject: [Asterisk-Users] Snom 360 - Multiple Server BLF Indications Hi, This is a weird request, but does anyone have a Snom 360 monitoring extensions for BLF on several Asterisk servers accross a network? Alternatively, can anyone give me a pointer as to how to setup a Snom 360 to monitor an extension not on it's own server? My scenario is that I have a main site which will have its own server (for storage of call recording data etc because the remote sites don't have the appropriate facilities) and each site has its own embedded system (to ensure that if the network goes down we can still use a normal telephone line). We need an operator telephone with expansion modules (hence the Snom 360) to monitor approximately 180 extensions on approximately 60 asterisk systems (about three extensions per site) so the operator can immediately see any extensions that successfully initiate a call. Any information would be greatly appreciated. Kind Regards Stuart -- -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 301 is slow
Now that's an interesting comment - most people think the speakerphone on the Polycom is quite good. Paul Hales Technical Manager AsteriskIT - Original Message - From: Denis Galvão - iSolve [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED]; Kevin P. Fleming [EMAIL PROTECTED] Sent: Monday, March 27, 2006 11:47 AM Subject: Re: [Asterisk-Users] Polycom IP 301 is slow The worst thing on all Polycom IP phones is the speaker phone's poor quality. You could not have a conference call using the speakers, only the head phone. Denis. On 26 de mar de 2006, at 21:17, Avi Miller wrote: Nick Hoffman wrote: Hrm, well that's disappointing. If they're so slow, why are they so popular? They may be slow to startup, but they're great phones. :) Once the phone has started up, it works like a charm and the sound/call quality is fantastic. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells)
AR Tarzi wrote: Not GSM/DECT but GSM/Wifi phones are available - This is not a recommendation, I don't like what I've seen. try www.imate.com (to start with) .. they have at least three types of GSM phones that do Wifi .. They run windows so there are several sip softwares and one IAX software that work with these - Also Nokia has a GSM phone that does Wifi but that's a symbian (OS) phone (don't know of sip software that works with it). The nokia E60 supports 'internet calls over WLAN' and lists a SIP API as one of it's device features (http://www.forum.nokia.com/main/0,,018-2715,00.html?model=E60). Actually, Nokia has had a SIP stack/SDK for the Series 60 for quite a long time. Just no actual UA to use it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 301 is slow
Denis Galvão - iSolve wrote: The worst thing on all Polycom IP phones is the speaker phone's poor quality. You could not have a conference call using the speakers, only the head phone. WHAT! The Polycom phones that have speaker phone features (the 50x/60x) are great speaker phones. The 301 is not an speaker phone. It only has a listen-only hands free setup. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 301 is slow
Nick Hoffman wrote: On Sat March 25 2006 18:06, Nick Hoffman [EMAIL PROTECTED] wrote: Hi guys, I've been using a Polycom IP 301 for a couple of weeks now and find that it's extremely slow for configuring. For instance, it takes several minutes to boot up, apply any changes via the web interface takes at least a minute, etc. Is this normal behaviour? Is there anything that can be done about it? On Sun March 26 2006 05:35, [EMAIL PROTECTED] wrote: polycoms are just that way. they are glacially slow on rebooting, and reboot for any trivial change. and no, nothing can be done about it. -Dan On Sun March 26 2006 06:20, Kevin P. Fleming [EMAIL PROTECTED] wrote: It is normal, and there is nothing you can do about it. The processor in the Polycom phones is not fast... Hrm, well that's disappointing. If they're so slow, why are they so popular? Because you don't normally reboot them. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Web based voicemail client
I'm looking for a good web based voicemail client that can use mysql or realtime drivers. I can't seem to get vmail.cgi to work with realtime. Thanks for any help you can give. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GSM/DECT handsets (was gsm picocells)
Not GSM/DECT but GSM/Wifi phones are available - This is not a recommendation, I don't like what I've seen. It strikes me as really strange that GSM/Wifi would be available while GSM/DECT is not so much. DECT is a voice technology, while wifi isn't. Still... there's a lot about the world I don't understand :) James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Web based voicemail client
Ari? Thanks, Steve Totaro http://www.asteriskhelpdesk.com -Original Message- From: Hall, Eric M. [mailto:[EMAIL PROTECTED] Sent: Sunday, March 26, 2006 9:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Web based voicemail client I'm looking for a good web based voicemail client that can use mysql or realtime drivers. I can't seem to get vmail.cgi to work with realtime. Thanks for any help you can give. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 301 is slow
Polycom is king as far as speakerphones IMHO. Thanks, Steve Totaro http://www.asteriskhelpdesk.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Sunday, March 26, 2006 9:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP 301 is slow Now that's an interesting comment - most people think the speakerphone on the Polycom is quite good. Paul Hales Technical Manager AsteriskIT - Original Message - From: Denis Galvão - iSolve [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED]; Kevin P. Fleming [EMAIL PROTECTED] Sent: Monday, March 27, 2006 11:47 AM Subject: Re: [Asterisk-Users] Polycom IP 301 is slow The worst thing on all Polycom IP phones is the speaker phone's poor quality. You could not have a conference call using the speakers, only the head phone. Denis. On 26 de mar de 2006, at 21:17, Avi Miller wrote: Nick Hoffman wrote: Hrm, well that's disappointing. If they're so slow, why are they so popular? They may be slow to startup, but they're great phones. :) Once the phone has started up, it works like a charm and the sound/call quality is fantastic. -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells)
Understanding..is not required. ;) PaulH - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 27, 2006 12:23 PM Subject: RE: [Asterisk-Users] GSM/DECT handsets (was gsm picocells) Not GSM/DECT but GSM/Wifi phones are available - This is not a recommendation, I don't like what I've seen. It strikes me as really strange that GSM/Wifi would be available while GSM/DECT is not so much. DECT is a voice technology, while wifi isn't. Still... there's a lot about the world I don't understand :) James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Web based voicemail client
If your talking about Asterisk Recording Interface this is what I found on the web site Submitted by dan.littlejohn on Wed, 12/28/2005 - 5:34am. ARI does not support realtime yet. It is coming Nice app but just can't do what I need it to. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sunday, March 26, 2006 9:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based voicemail client Ari? Thanks, Steve Totaro http://www.asteriskhelpdesk.com -Original Message- From: Hall, Eric M. [mailto:[EMAIL PROTECTED] Sent: Sunday, March 26, 2006 9:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Web based voicemail client I'm looking for a good web based voicemail client that can use mysql or realtime drivers. I can't seem to get vmail.cgi to work with realtime. Thanks for any help you can give. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] metermaid patch
Dr. Michael J. Chudobiak wrote: I'd like to be able to use my Snom 360 LEDs to view the status of parking slots, so I'm trying to install the metermaid patch (http://bugs.digium.com/view.php?id=5779). Can someone help an svn newbie figure out how to install this patch? I've done the following: Any update on this? Also, is there any chance that the metermaid functionality will be added to Asterisk? Matt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] help on mfc/r2
Well well, This is not right at all. You should have like 1001 for TxABCD and 1011 for RxABCD. That's why you are getting blocked. Something called my attention: Bipolar Violation: you have some of it, which confirms that there's something wrong. Indeed, your call file is ok. That's why I asked you about calling some extensions inside Alcatel box from a sip and/or iax2 phone. Question: if you have PRI, why are you using R2? Sugestion: check Alcatel's setup for this card. On Alcatel side, do you have any alarms or errors? BTW, do you use any kind of IM? []'s MM -Original Message- From: Krzysztof Drewicz [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Mon, 27 Mar 2006 02:49:47 +0200 Delivered: Sun, 26 Mar 2006 18:47:47 Subject:[Asterisk-Users] help on mfc/r2 Melcon Moraes napisaÅ(a): but can you make a simple call to your regular phone number from some SIP/IAX2 softphone/hardphone? Right now? It's not working. I could use (i have needed hardware). Data: SIP/extension-number Or even capi/zap channel. When e400p is configured as a E1/PRI and connected to diffrent card, my .call-file works (I hear asterisk-demo after receivng phone). So i assume my .call file is OK. What do you have in zaptel.conf file? For instance, I have this on my: span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 Me too. I've seen dchannel=16 somewhere on the net, but it makes me very confused, as R2 imvho is beeing digigit-digit on every litle time-slot/channel. hing to do with that. What happens if you use Unicall/g1 instead of just channel 1 of Unicall? Same, it's still: *CLI -- Attempting call on Unicall/g1/363 for application playback(demo-thanks) (Retry 1) Mar 27 01:42:00 NOTICE[13840]: channel.c:2437 __ast_request_and_dial: Unable to request channel Unicall/g1/363 Mar 27 01:42:00 NOTICE[13840]: pbx_spool.c:269 attempt_thread: Call failed to go through, reason 0 Do you have zttool compiled? How those bits are showed up? The timing â Current Alarms: No alarms. â â Sync Source: Tormenta 2 (PCI) Quad E1 Card â â IRQ Misses: 0 â â Bipolar Viol: 609 â â Tx/Rx Levels: 0/ 0 â â Total/Conf/Act: 31/ 30/ 30 â â 112333 ââââââââ â â 1234567890123456789012345789012 â Back â â â TxA 111 111 ââââââââ â â TxB 111 111 â â TxC 000 000 â â TxD 111 111 â â ââââââââ â â RxA 000 000 â Loop â â â RxB 111 111 ââââââââ â â RxC 000 000 â â RxD 111 111 â â â source is correctly set up? Seems that your master timing sync source is the Alcatel, so you should use span=1,1,0 ok, -- Krzysztof Drewicz [EMAIL PROTECTED] +48 22 34 54 363 E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,1143420467.988851.6696.balcomo.terra.com.br,6575,Des15,Des15 --Original Message Ends-- -- Melcon Moraes [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CHINA DID
Should be posted to the -biz list? Steve Ducat wrote: CHINA DID I am once again in search of China DID's. Either Shanghai (021) or Guangzhou (020). Please advise if you can supply. Steven Ducat. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dipura 2002 auto dial or intercom
Great! Thx a lot Paul, I guess this applies to all sipuras right? Anybody knows if this can also be done with polycoms? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Paul Hayes |Sent: Wednesday, March 15, 2006 5:34 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] dipura 2002 auto dial or intercom | |This called hot line or batphone (as it's like the phone |the commissioner used to have in Batman that went straight |through to Bruce Wayne). | |Set the dialplan to this: | |(S0:#) | |where is the number/SIP address you want to dial. |Note, that's a zero after the S. | | | |Anton Krall wrote: | |Guys. | |Anybody using sipuras 2002 knows if there is a way to make the phones |connected to it to autodial an extension when the phone is picked up? | |For example, if the phone is on a police booth (building |entrance) and you |want the guys to just pick up the phone and make the phone |auto dial the |receptionist extension without the guys having to dial anything (ala |batphone). | |Is this possible with spa's? | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Snom 360 problems
Release Notes for recent snom360 beta firmware: Release 5.5.1: o GUI: fixed consultative Xfer with fkeys Release 5.5: o GUI: fixed cursor handling (scrolling, backspace) in edit number state o GUI: put last active call on hold on top in holding/transfer Release 5.4: o GUI: added shared line LED blink when holding o SIP: fixed bug in ENUM lookup o LID: fixed port access for keep_alive where it could access a port that didn't exist anymore Release 5.3.6: o LID: made sure audio channels are off in idle mode under all scenarios Release 5.3.5: o GUI: added cwi ringer indication o GUI: fixed unnecessary dialog state switches on shared line offhook o GUI: status led for missed calls o SIP: RAck in PRACK was buggy o SIP: added call pickup for shared lines Release 5.3.4: o SIP: added +sip.rendering parameter for BLA hold/resume NOTIFYs o SIP: NOTIFYs with subscription-state: terminated remove the subscription Release 5.3.3: o GUI: fixed DND o GUI: fixed bug in displaying old voice mail messages o SIP: display local LED status for shared lines o WEB: + in settings value isn't anymore replaced by its hex value on settings dump web interface page o WEB: further enhanced french translation o SRTP: fixed bug with auto-answer Release 5.3.2: o GUI: setting_server can be set manually via GUI menu (snom360) o GUI: ringer device should not switch to speaker if headset is enabled o GUI: dkeys (e.g. Redial, Retrieve) are working in edit number state, too o SETTINGS: if setting_server is IP:port only, make a valid URL out of it o SIP: display local LED status for shared lines o SRTP: fix bug with auto-answer Release 5.3.1: o GUI: Shared Lines can be mapped to LEDs o LID: random number generated from random audio data Release 5.3: o GUI: blind-xfer via programmable keys doesnt require pressing the Enter key o GUI: incoming call context can be switched with the cursor o GUI: fixed freezing during calls on hold o GUI: added setting cancel_on_hold which, if set to false, makes the phone ignore any cancel key press in holding state o GUI: fixed DND, wasn't working properly after reboot during DND on o GUI: enhanced french translation o GUI: fixed, mute key stops working after 20 seconds if no DNS server is reachable o LID: further reduced ringer volumes o SIP: unsupported p-time values for codecs in responses disconnects the call o SIP: treat all return codes 100 and 180 as 180 Ringing o WEB: enhanced french translation - Usman Tahir snom technology AG -- Message: 13 Date: Sat, 25 Mar 2006 11:53:24 -0800 (PST) From: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: Snom 360 problems To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed On Fri, 24 Mar 2006, Usman Tahir wrote: For the conf on Xfer issue, use the latest beta http://fox.snom.com/download/snom360-5.5.1b-beta-SIP-j.bin what's the changelog for 5.5.1b? -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM/DECT handsets (was gsm picocells)
On 27/03/06, James Harper [EMAIL PROTECTED] wrote: Not GSM/DECT but GSM/Wifi phones are available - This is not a recommendation, I don't like what I've seen. It strikes me as really strange that GSM/Wifi would be available while GSM/DECT is not so much. DECT is a voice technology, while wifi isn't. 1) Because the phones do so much more than voice calls. Would you run a web browser over DECT, or would it work better over wi-fi? 2) How many public DECT hotspots do you know about? 3) How many companies have deployed DECT in their buildings? Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] stop monitor on transfer
Hi John, yes, Im using native transfer. What I do is use Monitor on the dialplan of the extension that picks up the call coming from PSTN, so after that, if the extension forward or transfers the call, monitor keeps recording all thru the end of the call no matter where it is been transferred to. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |John Daragon |Sent: Tuesday, March 21, 2006 4:54 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] stop monitor on transfer | |Anton Krall wrote: | Guys. | | This idea has been banging my headfor days now and I feel |the need to | share with you. | | Imagine this scenario: all calls come in thru a |receptionist, asterisk | records all incoming calls, the receptionist's work is to |transfer the | calls to internal people but some of them are bosses and you |know how | bosses are, they don't want their calls to be recorded, so, I have | been trying to figure a way on how to stop monitoring / |recoring calls | once they are transferred to a bosses extension while othe transferd | to other people stay on record mode. | |Anton, hi; | |I've got exactly the opposite problem. I *want* to record the |call after the transfer, but (using MixMonitor and SIP |transfers on Snom |handsets) the recording terminates with the transfer. | |Are you using Asterisk native transfer ? | |jd | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom batphone (was dipura 2002 auto dial or intercom)
To answer my own question :) Yes, it can be done by way of using polycoms magic config files: On phone1.cfg for all phones or MAC ADDRESS.cfg for a specific phone: call.autoOffHook.x.enabled=1 And call.autoOffHook.x.contact=EXTENSION TO CALL |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Anton Krall |Sent: Sunday, March 26, 2006 11:13 PM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] dipura 2002 auto dial or intercom | |Great! Thx a lot Paul, I guess this applies to all sipuras right? | |Anybody knows if this can also be done with polycoms? | ||-Original Message- ||From: [EMAIL PROTECTED] ||[mailto:[EMAIL PROTECTED] On Behalf Of Paul ||Hayes ||Sent: Wednesday, March 15, 2006 5:34 AM ||To: Asterisk Users Mailing List - Non-Commercial Discussion ||Subject: Re: [Asterisk-Users] dipura 2002 auto dial or intercom || ||This called hot line or batphone (as it's like the phone the ||commissioner used to have in Batman that went straight |through to Bruce ||Wayne). || ||Set the dialplan to this: || ||(S0:#) || ||where is the number/SIP address you want to dial. ||Note, that's a zero after the S. || || || ||Anton Krall wrote: || ||Guys. || ||Anybody using sipuras 2002 knows if there is a way to make |the phones ||connected to it to autodial an extension when the phone is picked up? || ||For example, if the phone is on a police booth (building ||entrance) and you ||want the guys to just pick up the phone and make the phone ||auto dial the ||receptionist extension without the guys having to dial anything (ala ||batphone). || ||Is this possible with spa's? || ||___ ||--Bandwidth and Colocation provided by Easynews.com -- || ||Asterisk-Users mailing list ||To UNSUBSCRIBE or update options visit: || http://lists.digium.com/mailman/listinfo/asterisk-users || || ||___ ||--Bandwidth and Colocation provided by Easynews.com -- || ||Asterisk-Users mailing list ||To UNSUBSCRIBE or update options visit: || http://lists.digium.com/mailman/listinfo/asterisk-users || | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom batphone (was dipura 2002 auto dial orintercom)
Man! I love those phones Great speakerphone, great functionality, works great with asterisk kiss ass mode Any polycom reps here? /kiss ass mode |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Anton Krall |Sent: Sunday, March 26, 2006 11:50 PM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: [Asterisk-Users] Polycom batphone (was dipura 2002 |auto dial orintercom) | |To answer my own question :) | |Yes, it can be done by way of using polycoms magic config files: | |On phone1.cfg for all phones or MAC ADDRESS.cfg for a specific phone: | |call.autoOffHook.x.enabled=1 |And |call.autoOffHook.x.contact=EXTENSION TO CALL | | ||-Original Message- ||From: [EMAIL PROTECTED] ||[mailto:[EMAIL PROTECTED] On Behalf Of Anton ||Krall ||Sent: Sunday, March 26, 2006 11:13 PM ||To: 'Asterisk Users Mailing List - Non-Commercial Discussion' ||Subject: RE: [Asterisk-Users] dipura 2002 auto dial or intercom || ||Great! Thx a lot Paul, I guess this applies to all sipuras right? || ||Anybody knows if this can also be done with polycoms? || |||-Original Message- |||From: [EMAIL PROTECTED] |||[mailto:[EMAIL PROTECTED] On Behalf Of Paul |||Hayes |||Sent: Wednesday, March 15, 2006 5:34 AM |||To: Asterisk Users Mailing List - Non-Commercial Discussion |||Subject: Re: [Asterisk-Users] dipura 2002 auto dial or intercom ||| |||This called hot line or batphone (as it's like the phone the |||commissioner used to have in Batman that went straight ||through to Bruce |||Wayne). ||| |||Set the dialplan to this: ||| |||(S0:#) ||| |||where is the number/SIP address you want to dial. |||Note, that's a zero after the S. ||| ||| ||| |||Anton Krall wrote: ||| |||Guys. ||| |||Anybody using sipuras 2002 knows if there is a way to make ||the phones |||connected to it to autodial an extension when the phone is |picked up? ||| |||For example, if the phone is on a police booth (building |||entrance) and you |||want the guys to just pick up the phone and make the phone |||auto dial the |||receptionist extension without the guys having to dial |anything (ala |||batphone). ||| |||Is this possible with spa's? ||| |||___ |||--Bandwidth and Colocation provided by Easynews.com -- ||| |||Asterisk-Users mailing list |||To UNSUBSCRIBE or update options visit: ||| http://lists.digium.com/mailman/listinfo/asterisk-users ||| ||| |||___ |||--Bandwidth and Colocation provided by Easynews.com -- ||| |||Asterisk-Users mailing list |||To UNSUBSCRIBE or update options visit: ||| http://lists.digium.com/mailman/listinfo/asterisk-users ||| || ||___ ||--Bandwidth and Colocation provided by Easynews.com -- || ||Asterisk-Users mailing list ||To UNSUBSCRIBE or update options visit: || http://lists.digium.com/mailman/listinfo/asterisk-users || || | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 301 is slow
Denis Galvão - iSolve wrote: The worst thing on all Polycom IP phones is the speaker phone's poor quality. You could not have a conference call using the speakers, only the head phone. Huh. Polycoms have the best speakerphone I've ever used on an IP phone. :) -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP trunk problem
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Marty, But with the same 128 bit upstream circuit, directly connecting the SJPhone the Stun server and using ulaw, everything is perfect. The problem comes when i am putting Asterisk in the picture. I have used SJ Phone softphone. His first codec choice is gsm. If you didn't change anything in SJ Phone settings, and your provider allows gsm, then softphone connects with gsm codec. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Pickupexten not working
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... i can confirm that this exist on 1.2.5, and the last time i said this, the original poster was supposed to file a bug on bugs.digium.com. OK. Can anybody else confirm this? I don't wona report it if it isn't bug. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Re: OT: Unblocking bloced CID
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... It's a toll free number. You can call it from anywhere and the costs of the call go on the callee not the caller. Thank you. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk add-ons upgrade
I have running Asterisk 1.2.5 with addons 1.2.1. on Fedora Core 4. I have installed ooh323 from 1.2.1 addons. How to upgrade addons to 1.2.2 version and install new ooh323 driver? Do I need to install addons 1.2.2 if I only need new ooh323 driver? Can I just untar addons, and run make clean; make; make install and then execute following cd asterisk-ooh323c ./configure make make install I always run clean install (of everything), but now that is not an option so I'm not sure what exactly I need to do. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: TAC Case Cisco 7960 Proxy address showing up in callerID
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... That's good to know... this only affects 8.2, right? As far as I know, yes. I have been using 7.5 and now I use 7.4 on 7940 and 7960 and I didn't have those issues. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Problem with Queue periodic announcemnets
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I have setup several queues for a customer. Their periodic announcement says please wait for the next available agent, or press * to leave a voicemail. This does not work when the message is playing. The message stops, but the user is left in the queue. Q-exit with * works the rest of the time fine. Has anyone seen this or know if it shoudl actually work differently? I have setup only one queue, and periodic announcement doesn't say to press * to leave voicemail. I'm using asterisk 1.2.5. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Which g729 codec to download for a P4?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Sorry for being a bit of a newbie here but I find the docs or README for downloading the G.729 codec from Digium are not as detailed as I would like or just don't really break down the different versions to a point that I am clear on which one to grab. The choices for 32bit are: drwxr-xr-x3 004096 Dec 05 00:21 athlon-xp drwxr-xr-x3 004096 Dec 05 00:21 c3 drwxr-xr-x3 004096 Dec 05 00:21 c3-2 drwxr-xr-x3 004096 Dec 05 00:21 i386 drwxr-xr-x3 004096 Dec 05 00:21 i586 drwxr-xr-x3 004096 Dec 05 00:21 i686 drwxr-xr-x3 004096 Dec 05 00:21 k6-3 drwxr-xr-x3 004096 Dec 05 00:21 pentium-m drwxr-xr-x3 004096 Dec 05 00:21 pentium3m drwxr-xr-x3 004096 Dec 05 00:21 pentium4m drwxr-xr-x3 004096 Dec 05 00:21 prescott I remember I head the same issue. I don't know why they don't put txt file with explanation?! Next time I need their codec I'll definitely write them that suggestion. Anyway, I have used i386 for P4. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: problems compiling zaptel on FC5
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... now, i just edited the Makefile that comes in zaptel directory to disable any usb, as i am not going to use any usb device in my asterisk, and it compiles and work ok. Hi Raul! Please send us what lines did you comment. Does it work with all versions of Zaptel? -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Free g729
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hello, I installed Asterisk from CVS on Redhat Linux 9 and working with chan_h323 module and g729/g723 free codecs too. Can you send us more information about this free g729 codecs? -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: making ooh323 authenticate gateway just like sip does
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Can someone tell me how I can configure ooh323.conf to accept call from h323 gateway (only the authorized h323 gateway) to my asterisk. Sorry, this is not answer to your question, but I need to ask you something. Are you using ooh323 from 1.2.1 or 1.2.2 add-ons? I'm using 1.2.1 and this is what has happened to me few times. Asterisk shows that I have 11 active calls and 11 active channels but really, none of them is active. And when this happens * hang's up. Do you know anything about this? pbx*CLI show channels Channel Location State Application(Data) SIP/302-30d5 [EMAIL PROTECTED]:3RingDial(OOH323/[EMAIL PROTECTED] SIP/302-a782 [EMAIL PROTECTED]:3RingDial(OOH323/[EMAIL PROTECTED] SIP/301-878b [EMAIL PROTECTED]:3RingDial(OOH323/[EMAIL PROTECTED] SIP/302-eb92 [EMAIL PROTECTED]:3RingDial(OOH323/[EMAIL PROTECTED] SIP/301-5535 [EMAIL PROTECTED]:3RingDial(OOH323/[EMAIL PROTECTED] SIP/302-eb56 [EMAIL PROTECTED]:3 RingDial(OOH323/[EMAIL PROTECTED]|6 SIP/302-0568 [EMAIL PROTECTED]:3RingDial(OOH323/[EMAIL PROTECTED] SIP/302-386b [EMAIL PROTECTED]:3RingDial(OOH323/[EMAIL PROTECTED] SIP/302-f2ac [EMAIL PROTECTED]:3RingDial(OOH323/[EMAIL PROTECTED] SIP/301-466f [EMAIL PROTECTED]:3RingDial(OOH323/[EMAIL PROTECTED] SIP/301-a154 (None) Up Bridged Call(OOH323/xxx.xxx.xxx.xxx 11 active channels 11 active calls -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Best GUI for basic HostedPBX service
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi, I'm looking for a web GUI to offer my end-users (Hosted PBX), and I thought I'd pick a few brains first. I'm not looking to configure the Asterisk server itself, VI works adequately for that. But I want to give Web access to as many of the following features: This is something I'm will need in few months. If you find anything, please let the group know. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: reload - restart
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi ! What is now the difference between a: reload - (cli command reload). restart - (I assume the application asterisk is restarted. o.k starting from new) sip reload - (cli command sip reload). Is sip reload part of the reload command ? Please confirm: Which is the correct command when adding a new SIP party in sip.conf. sip reload ? reload - reloads all conf files sip reload - reloads sip.conf file restart - I have never use it, guess it restarts Asterisk -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cisco 7960 - Have to press a menu button to dial
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... You have to set up a dialplan.xml file in your tftpboot directory for the phone to pull: DIALTEMPLATE TEMPLATE MATCH=9,59. Timeout=0/ TEMPLATE MATCH=9,29. Timeout=0/ TEMPLATE MATCH=9,832... Timeout=0/ TEMPLATE MATCH=9,713... Timeout=0/ TEMPLATE MATCH=9,281... Timeout=0/ TEMPLATE MATCH=9,903... Timeout=0/ TEMPLATE MATCH=\*500 Timeout=0/ TEMPLATE MATCH=\*54 Timeout=0/ TEMPLATE MATCH=\*55 Timeout=0/ TEMPLATE MATCH=\*69 Timeout=0/ TEMPLATE MATCH=*Timeout=5/ !-- Anything else -- /DIALTEMPLATE Hi Aaron! Can you tell me what , and \ stands for? . changes only one number from 0 to 9, right? * changes unlimited number of numbers from 0 to 9? -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Best GUI for basic HostedPBX service
Please stop send me email Best Regards, Mr.Peeramate Rochanasmita Project Manager/General Manager SIPphone (Thailand) Co., Ltd. 644/19 Moo 1 Klong Kum, Bung Kum Bangkok Thailand 10230 SIP No.100888 SIP Call Center No.888 Tel. 0 2690 3999 Fax. 0 2690 3535 Mobile. 0 1423 1423 Email : [EMAIL PROTECTED] MSN : [EMAIL PROTECTED] Website : www.sipphone.co.th -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Par?ina Sent: Monday, March 27, 2006 1:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: Best GUI for basic HostedPBX service In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi, I'm looking for a web GUI to offer my end-users (Hosted PBX), and I thought I'd pick a few brains first. I'm not looking to configure the Asterisk server itself, VI works adequately for that. But I want to give Web access to as many of the following features: This is something I'm will need in few months. If you find anything, please let the group know. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Cisco 7960 - Have to press a menu button to dial
Hi Aaron! Can you tell me what , and \ stands for? . changes only one number from 0 to 9, right? * changes unlimited number of numbers from 0 to 9? -- Tomislav Parcina tparcina#lama.hr Absolutely right :) \ escapes the next character, so if you wants *69 to go through immediately, you'd put \*69 so that the * gets recognized as a digit. , returns the dialtone sound. When my users hit 9, they like to hear the dialtone still so they know they're dialing outside. You got . and * right. Never put a 0 timeout on * or nothing else will work right. Hope that helps. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?
Actually I've got five, but the first one I have received around Xmas and I don't have these problems with it. I use spa3000 as FXO and the gsm gateway works seamlessly inbound, outbound, DISA, no annoying sounds, no DTMF problems. There is one problem however, the gateway does not transfer correctly the CID to the FXO(at least in my case) but this could be a sipura problem as well. Now, the other 4 seam to be a different model or something and one should be very careful ordering that thing since you never know which model you are going to receive. They are used with no-brand-name FXO/FXS ATAs but I don't think that the ATAs is the problem. Everything goes wrong when the gateway is tested as a dock-n-talk (dialing through it connected to one of the RJ11 with an ordinary phone set). First there is no DTMF recognition whatsoever, and second tha gateway does not sense the hangup and start making the noises. Hope Sam could solve the problem with the factory or exchange the goods with working ones. Benchev Outch... Four of them and not working... That hurts. How do you connect them to * ? As I'm using only one for me an X100P-FXO is sufficiant and seems to work as good as attaching a real anlog phone. Btw. I saw that www.voipsolutions.be is selling them also, but for 165.- euro On Sun, Mar 26, 2006 at 03:28:07PM +0200, Benchev wrote: Hi Adibar, Thanks very much for the answer. We are also struggling (with 4 of them :( ) and will let you know how the things develop, too, in case of success. Have a nice Sunday, Benchev Hi Benchev I'm still in contact with Sam, but currently no changes. The device is still in an unusable state for me, as it only allows one call, which results in wild-beeping on terminating the call. But I still hope, that Sam finds anywhere a tech-person who can hand me out the correct setup-information. As soon as I get it in a working state, I will let you know it ;-) Adibar On Sat, Mar 25, 2006 at 09:55:56PM +0200, Benchev wrote: Hi Adibar, Any success with the gsm gateway? I have exactly the same problem with units received this month. The codes given by Sam are not working... Please, let me know if you have discovered something. Thanks in advance, Benchev But these are the wrong instructions again. Same as those ones you sent me allready. I've got the small box for £60 The only reaction I get is if I press just *. Then the display changes to SET___. After that there is silence for about 15 seconds. Pressing any keys is only allowd up to four digits. So also the given password is to long for entering. After the 15 seconds or the four digits I get a busy-signal. No password-prompt, no LINE CON. Nada... Adibar On Sat, Mar 11, 2006 at 05:12:40AM +0800, Sam Tam wrote: Hello To solved the beeping problem you need to first enter the configuration mode. I .Entry into SETTING STATUS 1) Pick up the phone, press the button of 0 ** #; 2) Screen display: ?SETUP?; Input pass: input pass word: 332808 Then will display IMPUT CON you can change the box working mode . use the command *000100#0#for set defaut ,billing mode. *000100#1#for one long tone mode *000100#2#for long tone mode Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of adibar Sent: Saturday, March 11, 2006 4:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ? Hi Dan To connect the gsm-gateway I'm using a X100P which is sufficiant and which works. But I'm exeriencing problems with the device itself. The gateway acts like a POTS for me, but according a sales-representive of CT they do use it like a phone ?!? Dialing in works fine (beside the currently missing CID). Calls are taken from ZAP and are routed internaly. I can pickup, talk and... hangup. But after hanging up the gateway starts beeping like hell and requires a reset. Outgoing calls are even worse. I can dial, opposite takes the call, gateway hangs up and beeps again. Therefore... reset again. To be shure that it's neither * nor the X100P I tried the same with an old analog phone. Same result. So, you see. as I'm currently not ready to use it, I even could not think about tweaking, fine-tuning and codec-testing ;-) As soon as the tech-departement from CT comes (hopefully) back to me with a solution I'm the going to treat that box ;-) adibar On Fri, Mar 10, 2006 at 10:15:44PM +0300, [EMAIL PROTECTED] wrote: Hi
[Asterisk-Users] 7940 with Asterisk?
Title: 7940 with Asterisk? I just picked up a Cisco 7940 from an Auction and would like to use it on an Asterisk box. Can anyone give me a pointer where I should start so I can get it working? Skeeve ___ Skeeve Stevens, RHCE Email: [EMAIL PROTECTED] Website: www.skeeve.org - Telephone: (0414) 753 383 Address: P.O Box 1035, Epping, NSW, 1710, Australia eIntellego - [EMAIL PROTECTED] - www.eintellego.net ___ I'm a groove licked love child king of the verse Si vis pacem, para bellum ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free g729
Tomislav Parčina wrote: Can you send us more information about this free g729 codecs? There is no such thing as a 'free' G.729 - The DSP Group has claimed and defended the Patents they hold against the algorithm and process. Please do not use Asterisk/Digium related resources to exchange this information - They are the liable party as they provide a licensed version of G.729 from DSPg. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help on mfc/r2
Melcon Moraes napisał(a): Well well, Question: if you have PRI, why are you using R2? Sugestion: check Alcatel's setup for this card. On Alcatel side, do you have any alarms or errors? I would like to build simple IVR soulution, whitch allows me to do free B-channels. Now I could (easilly!!): connection goes from Telco-Alcatel-PRIIVRPRIAlcatelConsultant/Agent. I've to use Alcatel's CCD (in other words i'm stuck with Agents pluged as Phones, and call distibution and queues in PABX box. it's not my dectision to do that). But i'm using TWO channels for every transfer, don't know how to free the B-channels and pass the connection to my box. Maybe better solution is on Alcatel Z24/SLI16 (plain 24/16 analog plugs with RJ11) and some 12/24 port FXO cards in my IVR-Asterisk box? Can I use a flash-hook switching on FXO port in asterisk? BTW, do you use any kind of IM? Jabber, Skype, polish GG. :-D (the lucky/magic numbers will be send to you in PM). I could call you (PTSN), just say in what timezone you are living. kd, -- Krzysztof Drewicz [EMAIL PROTECTED] +48 22 34 54 363 begin:vcard fn:Krzysztof Drewicz n:Drewicz;Krzysztof adr;quoted-printable:;;Wybrze=C5=BCe Gdy=C5=84skie 6C;Warszawa;Mazowieckie;01-531;Polska email;internet:[EMAIL PROTECTED] tel;work:+48 22 34 54 363 tel;cell:+48 606 698 295 x-mozilla-html:TRUE url:http://www.citicom.pl version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] help on mfc/r2
My TZ is GMT-3. Just waiting for the lucky/magic numbers. []'s MM -Original Message- From: Krzysztof Drewicz [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Mon, 27 Mar 2006 09:39:52 +0200 Delivered: Mon, 27 Mar 2006 04:10:23 Subject:[Asterisk-Users] help on mfc/r2 Melcon Moraes napisa³(a): Well well, Question: if you have PRI, why are you using R2? Sugestion: check Alcatel's setup for this card. On Alcatel side, do you have any alarms or errors? I would like to build simple IVR soulution, whitch allows me to do free B-channels. Now I could (easilly!!): connection goes from Telco-Alcatel-PRIIVRPRIAlcatelConsultant/Agent. I've to use Alcatel's CCD (in other words i'm stuck with Agents pluged as Phones, and call distibution and queues in PABX box. it's not my dectision to do that). But i'm using TWO channels for every transfer, don't know how to free the B-channels and pass the connection to my box. Maybe better solution is on Alcatel Z24/SLI16 (plain 24/16 analog plugs with RJ11) and some 12/24 port FXO cards in my IVR-Asterisk box? Can I use a flash-hook switching on FXO port in asterisk? BTW, do you use any kind of IM? Jabber, Skype, polish GG. :-D (the lucky/magic numbers will be send to you in PM). I could call you (PTSN), just say in what timezone you are living. kd, -- Krzysztof Drewicz [EMAIL PROTECTED] +48 22 34 54 363 E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,1143445073.756963.12328.baladonia.terra.com.br,5440,Des15,Des15 --Original Message Ends-- -- Melcon Moraes [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
