Re: [Asterisk-Users] Caller ID ?
Hello Tom,
On 26-Aug-2005 7:50, Tom wrote:
Most of the time i can find answers to my questions on the wiki, google,
or searching the list now i am stuck .
I have a small * box at my house running 1.0.9 stable and a devlite kit.
Every thing is awesome VM, IVR, Echo canceling, and Meetme are all
working great.
Nice isn't it?
But on Incoming caller id i need to add a 9 as a prefix to make it
easier to return call from my cordless phone (cheap vtech phone). I have
tried to search the list and also google but i think i am searching of
the wrong thing. If i could get a kick in the right direction that would
be great.
This is what I came up with: (Watch out for linewraps on the second line.)
; Incoming on normal line
; Incoming on normal line
exten = ${EDN_MAIN},1,LookupCIDName(${CALLERIDNUM})
exten = ${EDN_MAIN},2,GotoIf($[$[${CALLERIDNUM} = ] |
$[${CALLERIDNUM} = CID withheld]]?5:3)
exten = ${EDN_MAIN},3,SetCIDNum(9${CALLERIDNUM})
exten = ${EDN_MAIN},4,SetVar(__NETWORK=KPN-Prive)
exten = ${EDN_MAIN},5,Goto(int-dest,${EDN_MAIN},1)
Stijn
--
Met Vriendelijke groet/Yours Sincerely
Stijn Jonker [EMAIL PROTECTED]
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[Asterisk-Users] UK Caller ID with TDM400P
Title: Message Has anyone here managed to get UK Caller ID (BT) working using a TDM400P card?. I've gotthe latest drivers from CVS, but can't find clarification if UK caller ID is supported and if so what the settings should be. Cheers Graham ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Working NFAS config w 411p anyone?
I finally figured out that echo directives and channel specific stuff needs to go between group and channel otherwise it didn't work or just gave weird results. I still have a problem with fax detection in terms of it turning off echo canceling. I have tried both, incoming, and everything in between. Seems like a way to turn it off (echo can) in the dial plan would be useful rather than having to answer the call with fax detect. Thanks, Shane -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Friday, August 26, 2005 12:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Working NFAS config w 411p anyone? Shane Burrell wrote: Does anyone have a working NFAS config for Zapata and zaptel for 2 NFAS trunks? First two DS1s on tg 1 and other two on tg2? I just setup two TE411Ps a few weeks ago, each with an NFAS group on it (one was two spans, the other three). I followed the documentation in the sample zapata.conf file and it worked fine... when I remembered that the 'logical span numbers' start at zero, not one :-) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re:TE110P EuroISDN dial out timing out
Try different entry in this parameter. In Italy mobiles start with 3, while public services with 1 and normal user numbers with 0. Using pridialplan=none, every number different from 0 was resulting in termination code 1, normally used for number never seen on the network. Ciao Mauro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Queues
Hi, I do have two questions regarding call queues: 1) How can I reach that waiting calls are also removed on removing the last agent listening to the queue. All I found is the switch to prevent new calls enter the queue after the last agent left. 2) Currently my queue does ring the agent after playing the you are first. How can I have the phone start ringing while the message is played? Elmar ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re:TE110P EuroISDN dial out timing out
Mauro Zanin wrote: Try different entry in this parameter. In Italy mobiles start with 3, while public services with 1 and normal user numbers with 0. Using pridialplan=none, every number different from 0 was resulting in termination code 1, normally used for number never seen on the network. Ciao Mauro Well, as I might not have noted or was unclear with is that it's not working dialing PSTN phone numbers that in turn are forwarded to mobile phone. Removing the forward it works ok so it shouldn't be related to how numbers are beeing sent to the PBX. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell 2850 anyone ...
[EMAIL PROTECTED] wrote: I just setup a Dell 1800, not a 2850, which is working awesome.. only had to disable USB, which realistically no-one on a phone system would care about anyways. Oh, really? Only if you're running a 2.6 kernel or using a zaptel card you don't need it. -- Andreas Sikkema bbned NV Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Maximum retries error.
I often get a Maximum retries error while making outgoing calls. Why does this happend? Most of the time a reload solves the problem, but not all the time? What to do? Aug 26 09:52:46 WARNING[6613]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Non-critical Response) Regards, Arne Morten. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fax codec problem
Hello, I have the following problem when I send fax to asterisk: -- Executing RxFAX(IAX2/[EMAIL PROTECTED]:4569/2, /var/spool/asterisk/fax/_1125039307.1.tif) in new stack 2005-08-26 06:55:09 NOTICE[30852]: channel.c:1317 ast_read: Dropping incompatible voice frame on IAX2/[EMAIL PROTECTED]:4569/2 of format slin since our native format has changed to gsm Reading around, I understood that I would have to add `allow=slin' to iax.conf, but this hasn't changed anything. The dis/allow part of my iax.conf is: [general] ... disallow=all allow=gsm allow=slin ... [1002] ... disallow=all allow=gsm allow=slin ... But this didn't change anything. Then, I was told that I should comment the `allow=gms' line. But this made things worse. Now, when I dial asterisk I get the following error: 2005-08-26 07:50:44 NOTICE[31232]: chan_iax2.c:5783 socket_read: Rejected connect attempt from 213.210.63.73, requested/capability 0x8/0xff0f incompatible with our capability 0xf840. I don't know much of how codecs work or how to configure asterisk, so please help Thank You, Daniel begin:vcard fn:Daniel Grad n:Grad;Daniel org:Think Digital adr:;;;Bucharest;;;Romania email;internet:[EMAIL PROTECTED] tel;cell:+40724891882 note;quoted-printable:YM: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] ICQ: 219628263=0D=0A= AIM: lunarul url:http://www.online-business.ro version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] About asterisk realtime
Hi, I intend to use asterisk realtime. I have test it with sip.conf and extension.conf. It works fine. Anyone already use it in practice. I am not sure about its stability for I got the code from the cvs head, not the stable version. Any advice and help will appreciated ! Best Regards, Gary Li DO YOU YAHOO!? 雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] About asterisk realtime
The next stable release, to be released any day now, will include realtime as far as I know. It works well. It would be nice to have a “stable” release with it included so you do not pick up other bugs from CVS Head. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary Li Sent: Friday, August 26, 2005 2:56 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] About asterisk realtime Hi, I intend to use asterisk realtime. I have test it with sip.conf and extension.conf. It works fine. Anyone already use it in practice. I am not sure about its stability for I got the code from the cvs head, not the stable version. Any advice and help will appreciated ! Best Regards, Gary Li DO YOU YAHOO!? 雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum retries error.
Hi, I'm very interested to understand that Warning too..does it happen every 30 minutes?? g Arne Morten Johansen wrote: I often get a Maximum retries error while making outgoing calls. Why does this happend? Most of the time a reload solves the problem, but not all the time? What to do? Aug 26 09:52:46 WARNING[6613]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Non-critical Response) Regards, Arne Morten. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re:TE110P EuroISDN dial out timing out
Mauro Zanin wrote: Try different entry in this parameter. In Italy mobiles start with 3, while public services with 1 and normal user numbers with 0. Using pridialplan=none, every number different from 0 was resulting in termination code 1, normally used for number never seen on the network. What about pridialplan=unknown ? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re:TE110P EuroISDN dial out timing out
Eric Wieling aka ManxPower wrote: What about pridialplan=unknown ? As noted before, it's not reaching the end number that's the problem as what pridialplan would help with sending the correct numbers out. Dialing to a non forwarded phone works, but if you forward that phone to a mobile phone it breaks. Looking at the PBX end everything seems fine (as in what numbers are beeing sent etc) but then asterisk suddenly sends a notice about not getting an answer. But I've tried the pridialplan settings just to be sure without success. Thanks anyway. :) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared
This is an interesting document about VoIP and Echo. http://www.cisco.com/univercd/cc/td/doc/cisintwk/intsolns/voipsol/ea_isd.htm ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Maximum retries error.
There is no static interval. But i found out that it was my IP-Phone Service Provider that was having serviceproblems today. -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Giorgio Incantalupo Sendt: 26. august 2005 11:33 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] Maximum retries error. Hi, I'm very interested to understand that Warning too..does it happen every 30 minutes?? g Arne Morten Johansen wrote: I often get a Maximum retries error while making outgoing calls. Why does this happend? Most of the time a reload solves the problem, but not all the time? What to do? Aug 26 09:52:46 WARNING[6613]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Non-critical Response) Regards, Arne Morten. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Updated Patch to chan_agent.c for PREACKANNOUNCE
Hmmm. I am often surprised when I don't get a response to a post that I think would interest at least _one_ person in the community. This one surprised me a little more, since I offered some code ;-). This morning, I just got a bounce notice that it was undelivered, which might explain it, except that I received the original post back through the list, so I don't understand it at all... Anyway, I solved the one bone-headed problem that I describe below, namely why did the agents show up in one DB and not the other. I didn't set the persistent keyword in the agents.conf file (doh...). All of my other questions still apply, as well as my offer to share the code/patch. Original Message Subject: [Asterisk-Users] Updated Patch to chan_agent.c for PREACKANNOUNCE Date: Thu, 18 Aug 2005 16:28:19 -0400 From: Hadar Pedhazur [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com First, many thanks to Greg Boehnlein for his patch to chan_agent.c for adding a preackannounce option. I am running CVS HEAD from 2005/07/31, and the patch failed in a few hunks, since the code was refactored to add in some CASE statements where there were compound if statements before. Anyway, I have successfully updated the patch to work against head as of 3 weeks ago, and would happily share that with anyone who is interested (just drop me a line off list). If a diff is preferable to the full 70k of C, just let me know what the correct options are for creating a diff suitable for patching the asterisk tree. OK, that said, I have a few questions and comments on this topic. This is my first use of the Queue command (very successfully so far), but I am afraid that expanding my use will require further patches, and I would like to verify that first. 1) If I use the syntax: Member = SIP/100 (rather than member = Agent/100, which maps to SIP/100) Then ackcall isn't used at all. In other words, a hard-wired member seems to ignore the agents.conf file completely. Is this the desired behavior? (It isn't for me...) 2) Since agents.conf is a separate file from queues.conf, having multiple queues does _not_ permit multiple preackannounce messages, each tied to a different queue (this strikes me as having better been patched into the Queue command). Similarly, you can't have one queue that has ackcall=yes, and another with ackcall=no. 3) I have the _exact_ same source version of CVS HEAD (from 2005/07/31) running on different servers (after a cvs co, I tar the source so that I can be sure I'm running _identical_ versions). On one machine, when an Agent logs in, I can see it in the DB, database show shows a key of: //Agents/1001 : [EMAIL PROTECTED];1001 On another machine, the DB shows _nothing_, yet the AgentCallbackLogin application works correctly (logging agents in and out), and shows the correct mapping on the CLI during a login. Still, the DB has _no trace_ of the Agents. I can't explain the difference in behavior, and would _love_ to have someone solve that mystery for me. I'm hoping that I am missing something obvious in the interaction between the Queue command and the Agents channel, and that some kind soul here will educate me. Otherwise, I think I might be off to doing more work in C than I ever though I would again in my life ;-). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk: Unable to read password.
Hello,
I am using asterisk as voicemail for my sip proxy.
When a user (1234)dials , the call is forwarded to
asterisk. However I receive the following error:
--Executing VoiceMailMain(SIP/1234-9afc, 1234) in
new stack
--Playing 'vm-password' (language 'en')
[WARNING]: app_voicemail.c:3359 vm-execmain: Unable to
read password
==Spawn extension (default, , 1) exited non-zero
on 'SIP/1234-9afc'
My configs are as follows:
;sip.conf
[1234]
type=friend
host=dynamic
context=default
mailbox=1234
;extensions.conf
[default]
exten=1234, 1, Voicemail(u${EXTEN})
exten=1234, 2, Hangup
exten=, 1, VoicemailMain(${CALLERIDNUM})
;voicemail.conf
1234=1234, P, [EMAIL PROTECTED]
Please advise if possible as i have looked through the
asterisk mail archives but cannot see what would be
wrong with the configuration.
many thanks.
___
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[Asterisk-Users] bridging sip to capi, no playtones back to caller
I've the following setup :
sip phone - ser (auth and routing) - asterisk with capi isdn
when I call a pstn number everything works fine, but I can't hear
anything till the called answer.
this is the output from a test call :
-- Executing Playtones(SIP/2.7.184.61-08152880, dial) in new stack
-- Executing Dial(SIP/2.7.184.61-08152880,
CAPI/02myisdnnum:347callednum) in new stack
-- creating pipe for PLCI=-1
sent CONNECT_REQ MN =0x193
-- Called 02myisdnnum:347callednum
-- CAPI[contr1/02myisdnnum]/2 is making progress passing it to
SIP/2.7.184.61-08152880
-- CAPI[contr1/02myisdnnum]/2 is ringing
sent FACILITY_REQ (PLCI=0x101)
-- CAPI[contr1/02myisdnnum]/2 answered
== Spawn extension (default, 347callednum, 2) exited non-zero on
'SIP/2.7.184.61-08152880'
asterisk-pri-1:/etc/asterisk # cat extensions.conf
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten = _X.,1,Playtones(ring)
exten = _X.,2,Dial,CAPI/0226265583:${EXTEN}
exten = _X.,3,HangupSIP/2.7.184.61-08152880
-- CAPI Hangingup
sent DISCONNECT_B3_REQ NCCI=0x10101
sent DISCONNECT_REQ PLCI=0x101
-- removed pipe for PLCI = 0x101
asterisk-pri-1:/etc/asterisk # cat sip.conf
[general]
context=default
port=5060
bindaddr=192.168.1.101
srvlookup=no
canreinvite=no
disallow=all
allow=alaw
asterisk-pri-1:/etc/asterisk # cat capi.conf
[general]
nationalprefix=0
internationalprefix=0039
rxgain=0.8
txgain=0.8
[interfaces]
msn=02myisdnnumber
incomingmsn=*
controller=1
softdtmf=0
context=default
callgroup=1
mode=immediate
devices=2
asterisk-pri-1:/etc/asterisk # cat indications.conf
[general]
country=it
[it]
description = Italy
ringcadence = 1000,4000
dial = 425/600,0/1000,425/200,0/200
busy = 425/500,0/500
ring = 425/1000,0/4000
congestion = 425/200,0/200
callwaiting = 425/200,0/600,425/200,0/1
dialrecall = 470/400,425/400
record = 1400/400,0/15000
info =
!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
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[Asterisk-Users] IAX2 Softphone Quality Network Cards
We are in the process of an Asterisk call center deployment using IAX2 G711 ulaw softphones. Outbound sound quality is terrible. This week we rebuilt the entire LAN with Cisco 2950-EI switches and have employed QoS on the switches and router. Still sounds terrible. What we are now finding is that the network card in the PC may be the key to the problem. A Dell Optiplex P4 2.4GHz 512MB machine with an onboard Intel NIC is bad, while an older Dell Dimension P3 864MHz 128MB machine with onboard 3COM sounds good. Has anyone out there had a similar experience? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Optipoint 600 Cant boot - development shell active
Hi, The only thing i know is that you need a netbootserver using five special files. So, if possible, ask Siemens for the optipoint 600 netboot upgrade procedure. AFAIK it is a known problem... hope it helps... --- Anthony Cox [EMAIL PROTECTED] wrote: Not strictly a problem with Asterisk but one of my phones. A couple of days ago I decided to update the firmware in my Optipoint 600 Office which looked as though it went swimmingly until that is, it rebooted. Since then the phone just boots up and displays the following: Can't Boot!! Development shell active. It doesn't try to request a DHCP address, in fact it does seem to do anything on the network and the key pad does nothing. Can anyone suggest a remedy? Anyone know how to get the development shell to do something? Thanks in advance. Anthony. -- Anthony Cox ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] system crash
We just had * crash on us - no calls could be made / received. We had to kill -9 the * process. Checking the error logs, I came across these two lines, with the times matching the crash: Aug 26 13:48:00 WARNING[19282] pbx.c: Local/[EMAIL PROTECTED],2 already has PBX structure?? Aug 26 13:48:00 WARNING[19282] channel.c: Thread -1105359952 Blocking 'Local/[EMAIL PROTECTED],2', already blocked by thread -1105626192 in procedure ast_waitfor_nandfds anyone got any clues on this (has it been fixed recently ?). We are running cvs-head as of 2 months ago. We are in the process of moving to cvs-head (or 1.2), so it's not a major for us, unless this is a bug still sitting in cvs-head. Julian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CD copy
Hi, I have 2 CDs that would like to make a backup of , I am having a hard time doing. I have tried NERO ver.6 but it does not work it always report unrecoverable sector. Does anyone knows of a copy tools to use to copy the CD Any help will be very nice and appreciated. Thank you all... Ellafi Yahoo! Mail for Mobile Take Yahoo! Mail with you! Check email on your mobile phone.___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] realtime sip channel configuration - insecure option
Hi all
I'm trying to figure out what values are valid for the insecure option in a
realtime configuration table. The table field is 4 chars long and the actual
valid values for this is longer. Can I modify the field length or has this
changed? Below is where I looked, if I'm not looking in the right place
please let me know.
the field on the table is:
...
`insecure` varchar(4) default NULL,
...
(http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Sip)
the actual values for this option (that I have found) are:
port: ignore the port number where authentication came from
invite: don't require initial INVITE to authenticate
port,invite: don't require initial INVITE to authenticate and ignore the port
where the request came from
(http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+insecure)
also found this on chan_sip.c:
/*--- insecure2str: Convert Insecure setting to printable string ---*/
static const char *insecure2str(int port, int invite)
{
if (port invite)
return port,invite;
else if (port)
return port;
else if (invite)
return invite;
else
return no;
}
thanks
Guillermo Krepper
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Re: [Asterisk-Users] Re: Asterisk 1.0.9 on SuSE 9.2 with ISDN BRI zaphfc?
Alessio Focardi ha scritto: Hello Lars, Have you got kernel sources installed ? I think that are mandatory for Zaphfc. Regards Not only, you have to have the kernel config save file too. Remember to make dep too. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voip-info - is it alive
I cannot reach voip-info - is it just me or is the site not available ? Julian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CD copy
Ellafi Fituri wrote: */Hi,/* */I have 2 CDs that would like to make a backup of , I am having a hard time doing. I have tried NERO ver.6 but it does not work it always report unrecoverable sector./* */Does anyone knows of a copy tools to use to copy the CD/* */Any help will be very nice and appreciated. Thank you all.../* */Ellafi/* Just plug an external drive into a USB port on your * server. There are several GUI and command line tools available. See the linux documentation project for help or use google. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CD copy
On Fri, Aug 26, 2005 at 06:42:33AM -0700, Ellafi Fituri wrote: Hi, I have 2 CDs that would like to make a backup of , I am having a hard time doing. I have tried NERO ver.6 but it does not work it always report unrecoverable sector. Nero? what is it? I don't have it in my apt repository ;-) Does anyone knows of a copy tools to use to copy the CD dd if=/dev/cdrom of=cd.iso For better handling of errors, maybe use ddrescue , http://packages.debian.org/ddrescue -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared
Hi, I'm not the OP, but I had a similar problem, in my case fxotune ran successfully for just one out of 3x FXO modules, but the coefficients were all 0's. My kernel is 2.6.11 on CentOS 4.1. So I'm curious if 2.6 kernel (instead of 2.4) has any input in this whole echo issue, not just fxotune. Yesterday I switched to KB1 echo canceller, it is by far the best. But today I had a similar experience to Eric Rees's Strange Echo post. After transfering to another internal line, echo starts. My theory is that after transfer some characteristics of the internal connection change, especially the Tx voice (the person talking on our side changes). So if the echo canceller is too committed to the voice of the first person answering the line (the operator), that would be quite expected. I don't know how KB1 or other echo cancellers work, but if I'm right, it would be better if echo canceller readjusted itself after transfer. Sorry if that's plain wrong. Can somebody comment please? I'm really interested in all posts in this thread and others or documents on echo. Btw, thanks Eric Wieling for the Cisco link. Soner - Original Message - From: Derek Whitten [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, August 26, 2005 2:39 AM Subject: Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tools for Remote Monitoring and User Management
At present, I would recommend [EMAIL PROTECTED] It comes with some monitoring tools as well as AMP. Darren Wiebe [EMAIL PROTECTED] Zeeshan Zakaria wrote: Hi all, What are the best and free tools for remotely adding, removing users on Asterisk server and also for monitoring the status of the Asterisk server, like how many users are logged on etc. I need tools for which I don’t have to pay. Thanks, Zeeshan A Zakaria www.acabling.com http://www.acabling.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type
Hi, is there anybody who knows what this warning means?? WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type TIA Giorgio -- GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bridging sip to capi, no playtones back to caller
On Fri, 26 Aug 2005, Simone Cittadini wrote:
I've the following setup :
sip phone - ser (auth and routing) - asterisk with capi isdn
when I call a pstn number everything works fine, but I can't hear anything
till the called answer.
If you want tones from isdn before the connection is established, you need
to set 'early-B3'. With older chan_capi versions, you need to put
'b' or 'B' at the beginning of your 'callednum'. See README of chan_capi.
If you want to use newer chan_capi, have a look at sourceforge.net.
Armin
this is the output from a test call :
-- Executing Playtones(SIP/2.7.184.61-08152880, dial) in new stack
-- Executing Dial(SIP/2.7.184.61-08152880,
CAPI/02myisdnnum:347callednum) in new stack
-- creating pipe for PLCI=-1
sent CONNECT_REQ MN =0x193
-- Called 02myisdnnum:347callednum
-- CAPI[contr1/02myisdnnum]/2 is making progress passing it to
SIP/2.7.184.61-08152880
-- CAPI[contr1/02myisdnnum]/2 is ringing
sent FACILITY_REQ (PLCI=0x101)
-- CAPI[contr1/02myisdnnum]/2 answered
== Spawn extension (default, 347callednum, 2) exited non-zero on
'SIP/2.7.184.61-08152880'
asterisk-pri-1:/etc/asterisk # cat extensions.conf
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten = _X.,1,Playtones(ring)
exten = _X.,2,Dial,CAPI/0226265583:${EXTEN}
exten = _X.,3,HangupSIP/2.7.184.61-08152880
-- CAPI Hangingup
sent DISCONNECT_B3_REQ NCCI=0x10101
sent DISCONNECT_REQ PLCI=0x101
-- removed pipe for PLCI = 0x101
asterisk-pri-1:/etc/asterisk # cat sip.conf
[general]
context=default
port=5060
bindaddr=192.168.1.101
srvlookup=no
canreinvite=no
disallow=all
allow=alaw
asterisk-pri-1:/etc/asterisk # cat capi.conf
[general]
nationalprefix=0
internationalprefix=0039
rxgain=0.8
txgain=0.8
[interfaces]
msn=02myisdnnumber
incomingmsn=*
controller=1
softdtmf=0
context=default
callgroup=1
mode=immediate
devices=2
asterisk-pri-1:/etc/asterisk # cat indications.conf
[general]
country=it
[it]
description = Italy
ringcadence = 1000,4000
dial = 425/600,0/1000,425/200,0/200
busy = 425/500,0/500
ring = 425/1000,0/4000
congestion = 425/200,0/200
callwaiting = 425/200,0/600,425/200,0/1
dialrecall = 470/400,425/400
record = 1400/400,0/15000
info =
!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
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Re: [Asterisk-Users] Simple Fax question
T.38 isn't a trivial enhancement, and I think that the community should consider itself extremely fortunate if someone actually gets T.38 implemented (including DSPs) for as little as $5500 being the motivation. True, but Steve Underwood does already has a lot of the DSP stuff done already with spandsp, doesn't he? Surely that is the hard part, right? Steve, are you currently working on t.38? Can a lot of the spandsp code be used? Are you getting/interested in getting financial support to work on this? I'm sure we could get all kinds of support for a project like this from the more commercial members of the community. Maybe there just needs to be more organization on the part of the people who want it done? Terry ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip-info - is it alive
I've been trying for 18 hours ... ;) Julian Giorgio Incantalupo wrote: Hi, sometimes it is not available. Be patient, wait 10 minutes and try again. Giorgio Julian Lyndon-Smith wrote: I cannot reach voip-info - is it just me or is the site not available ? Julian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple Fax question
True, but Steve Underwood does already has a lot of the DSP stuff done already with spandsp, doesn't he? I sincerely apologize for that first sentence... wow. :-) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Softphone Quality Network Cards
I haven't had similiar experience, but in several threads about sound quality people have talked about Network cards being the culprit. In particular, a few people have commented all sorts of problems on onboard NIC's, since they tend to be of lesser quality than stand-alone NICS. On 8/26/05, Adam Robins [EMAIL PROTECTED] wrote: We are in the process of an Asterisk call center deployment using IAX2 G711 ulaw softphones. Outbound sound quality is terrible. This week we rebuilt the entire LAN with Cisco 2950-EI switches and have employed QoS on the switches and router. Still sounds terrible. What we are now finding is that the network card in the PC may be the key to the problem. A Dell Optiplex P4 2.4GHz 512MB machine with an onboard Intel NIC is bad, while an older Dell Dimension P3 864MHz 128MB machine with onboard 3COM sounds good. Has anyone out there had a similar experience? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cvs update error?
Hi, Im experiencing a problem with playing back my voicemail. (Failed to write frame). It has been indicated in the archives that this is problem can be solved by updating asterisk from the cvs. I did make update in the /usr/src//asterisk directory to resolve this. However I got a message saying The following files have conflicts: channels/MakeFileCould someone advise me on what I need to do now to resolve these issues? Many thanks. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voip-info - is it alive
I have had no trouble reaching it. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Friday, August 26, 2005 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voip-info - is it alive I've been trying for 18 hours ... ;) Julian Giorgio Incantalupo wrote: Hi, sometimes it is not available. Be patient, wait 10 minutes and try again. Giorgio Julian Lyndon-Smith wrote: I cannot reach voip-info - is it just me or is the site not available ? Julian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk wiht LDAP
I am trying to configuring/running Asterisk::LDAP perl module getting from http://projects.alkaloid.net/ but no luck i have successfully installed this module but when i include its scheme file which is asterisk.scheme in the LDAP include list and try to start the LDAP Server service its gives the following error: /etc/init.d/ldap start Starting ldap-server/etc/openldap/schema/asterisk.schema: line 181: Unexpected token before 1.3.6.1.4.1.1466.115.121.1.36 EQUALITY numericStringMatch ) ObjectClassDescription = ( whsp numericoid whsp ; ObjectClass identifier [ NAME qdescrs ] [ DESC qdstring ] [ OBSOLETE whsp ] [ SUP oids ]; Superior ObjectClasses [ ( ABSTRACT / STRUCTURAL / AUXILIARY ) whsp ] ; default structural [ MUST oids ] ; AttributeTypes [ MAY oids ]; AttributeTypes whsp ) startproc: exit status of parent of /usr/lib/openldap/slapd: 1 failed Its include path is : /etc/openldap/schema/asterisk.schema But when i comment this line from LDAP Server started successfully can anyone trying this app kindly helping me out ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk wiht LDAP
I am trying to configuring/running Asterisk::LDAP perl module getting from http://projects.alkaloid.net/ but no luck i have successfully installed this module but when i include its scheme file which is asterisk.scheme in the LDAP include list and try to start the LDAP Server service its gives the following error: /etc/init.d/ldap start Starting ldap-server/etc/openldap/schema/asterisk.schema: line 181: Unexpected token before 1.3.6.1.4.1.1466.115.121.1.36 EQUALITY numericStringMatch ) ObjectClassDescription = ( whsp numericoid whsp ; ObjectClass identifier [ NAME qdescrs ] [ DESC qdstring ] [ OBSOLETE whsp ] [ SUP oids ]; Superior ObjectClasses [ ( ABSTRACT / STRUCTURAL / AUXILIARY ) whsp ] ; default structural [ MUST oids ] ; AttributeTypes [ MAY oids ]; AttributeTypes whsp ) startproc: exit status of parent of /usr/lib/openldap/slapd: 1 failed Its include path is : /etc/openldap/schema/asterisk.schema But when i comment this line from LDAP Server started successfully can anyone trying this app kindly helping me out ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Benchmarking / Stress Testing
Anyone have a good tool(s) to use for simulating a bunch of calls? Benchmarking or stress testing? I only need SIP protocol, and do appreciate any replies...I realize I could google it, but I am looking for opinions as well. Sherwood McGowan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT - Packet 8 firmware
A little off topic but for packet 8 users out there using asterisk behind a x100p check out the new firmware http://www.dslreports.com/r0/download/872826~6e5c593b26b72aef4bf68f6710eed5b8/sip1315unl.zip Allows you to assign your own codecs, currently using g711 90kbs and sounds amazing in comparison to the g729 30kbs thats standard. If you have the bandwidth to burn its a great idea. Cheers, Dean Use at your own risk blah blah blah ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Tools for Remote Monitoring and User Management
ARTCP (not yet released) will be doing exactly this, along with Zabbix for monitoring (custom UserParameters will be included in ARTCP) for *REALTIME. --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -Darren Wiebe -Sent: Friday, August 26, 2005 10:23 AM -To: [EMAIL PROTECTED]; Asterisk Users Mailing List - -Non-Commercial Discussion -Subject: Re: [Asterisk-Users] Tools for Remote Monitoring and -User Management - -At present, I would recommend [EMAIL PROTECTED] It comes with -some monitoring tools as well as AMP. - -Darren Wiebe -[EMAIL PROTECTED] - -Zeeshan Zakaria wrote: - - Hi all, - - What are the best and free tools for remotely adding, -removing users - on Asterisk server and also for monitoring the status of -the Asterisk - server, like how many users are logged on etc. I need tools -for which - I don't have to pay. - - Thanks, - - Zeeshan A Zakaria - - www.acabling.com http://www.acabling.com/ - -- --- -- - -___ ---Bandwidth and Colocation sponsored by Easynews.com -- - -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - - -___ ---Bandwidth and Colocation sponsored by Easynews.com -- - -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type
Giorgio Incantalupo wrote: Hi, is there anybody who knows what this warning means?? WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type type = friend type = user type = peer TIA Giorgio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fedora core 3 kernel source - could someone throw the dog a bone!
Ok, dont flame me, I know this is a question with an obvious answer to some, but I am not one of them. Installed FC3, but this time I decide to update since my ISOs are a bit old, so typical yum update Downloaded the FC3 SRPM for my kernel 2.6.12 Installed the SRPM package Ran rpmbuild bp target=i686 kernel-2.6.spec Tried to build zaptel error; You do not appear to have the sources for the 2.6.12-1.1372_FC3smp kernel installed. So I assume that either a) I did not build the correct source for the smp kernel, or b) I am missing a symbolic link to the kernel source. No help from the FC3 release notes, no help from a Google. So, if you dont mind, throw me the bone ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Replace Aspect by using Asterisk
Hi All: We have a very old Aspect ACD in our call center, I am doing research to replace it by using Asterisk, my boss has some kind of questions about capability and reliability of Asterisk, does anyone have done this kind of work with good result? I need some examples to convince him. Thanks, Tielin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can exsiting router handle VoIP traffic?
Short answer: Yes. It's just data. Long answer: In your LAN: Usually depends on the nature of the other data on your LAN. If you LAN has a ton of traffic you will have to use something like QoS tagging to ensure that your voice traffic is prioritized. Any decent switch supports this tagging and / or will retransmit frames as-recieved. In a lot of call centres you might see the agents using telnet sessions or a web based CRM which is lightweight traffic - wise and won't interfere with VoIP even without QoS, except in extreme cases where you have thousands of users in multiple subnets with single bottlenecks like everyone hitting the same server. On the Internet: Because the Internet is best effort by design there's no guarantee that packets will be delivered in-order, out-of-order, or even at all. Any quality of service tagging you do on your end is largely a pointless exercise because intermediate routers between you and your service provider will not honor the tag. As well, an inherent risk for a pure VoIP setup on the Internet is DoS'ing - a single script kiddie can make your day bad. Consider what would happen to your call centre should another Code Red day happen. You have to make a business decision as to whether the cost savings and flexibility that a pure VoIP setup would give you vs the risk of the call centre being without service for X amount of hours because of Internet problems. You may want to consider a hybrid approach, where you under-provision a PSTN connection such as a PRI (say, a single PRI for a couple of hundred users) and have calls overflow to a VoIP provider once the channel limit is reached. This way, you can take advantage of some of the cost savings and flexibility of VoIP and you have a backup that automatically kicks in should your Internet connection or VoIP provider goes down. If this happens, your capacity to process calls is diminished, but not completely toast. It's nice in Asterisk, because you *can* do this as opposed to a lot of other PBX'es where it's their way or the highway. One last thing to consider: I see that you work for Nintendo. It's my understanding that the latest Ethereal builds can identify and decode SIP and IAX packets to audio. What would Nintendo's feelings be on call centre data being transmitted on the Internet where it would be possible to intercept and decode this data. What would the legal and / or corporate ramifications be if this did happen? You can argue that this kind of thing can happen with a PSTN connection, however, that requires physical proximity and access to the line itself, which you control. In a VoIP scenario, once the packets traverse your firewall and get onto the Internet, they are essentially public property. hth -Original Message- From: Tielin Xu [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 24, 2005 10:22 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Can exsiting router handle VoIP traffic? Hi All: I'd like to test a pure VoIP call center set up under Asterisk, Can I use existing IP routers to get VoIP traffic from service provider to Asterisk with good quality of voice? In other words, do I have to do any hardware upgrade to make VoIP work in existing enterprise environement, we have 10g Ethernet LAN? Many thanks, Tielin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fedora core 3 kernel source - could someone throw the dog a bone!
This could be a duplicate post, sent it originally 4 hours ago, it never showed up! I know this is a question with an obvious answer to some, but I am not one of them. Installed FC3, but this time I decide to update since my ISOs are a bit old, so typical yum update Downloaded the FC3 SRPM for my kernel 2.6.12 Installed the SRPM package Ran rpmbuild bp target=i686 kernel-2.6.spec Tried to build zaptel error; You do not appear to have the sources for the 2.6.12-1.1372_FC3smp kernel installed. So I assume that either a) I did not build the correct source for the smp kernel, or b) I am missing a symbolic link to the kernel source. No help from the FC3 release notes, no help from a Google. So, if you dont mind, throw me the bone ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] updating display of a hardphone based on agents logging in
I've been thinking about how one would accomplish the same thing. I've got a CTI enabled GUI that tells the agent that they're logged in with the call centers that I've deployed thus far, but it's not quite the same as the agent just being able to look at the phone as well and know that they're logged in or not. You've got the Snom 320's, so maybe the most straight forward thing to do would be to use the Hint application with them to light a status LED when an agent is logged in and have it go dark when the agent is logged out. Are you using AgentCallBack? I wonder if Hint could be used to status the agent channel itself. Hmmm. Will have to check this out a little bit more. :-) On 8/25/05, Franklin Webb [EMAIL PROTECTED] wrote: Greetings all, We are settng up a fair sized call center on Asterisk, but we are having some issues with our agents not knowing if they have logged in and logged out. Prior to beginning our migration to VoIP the agents logged into our nortel phones and confirmation was displayed on the phone. My question is has anyone out there done anything from Asterisk that can change the display on a VoIP hardphone? We are currently using the Snom 320 and the Aastra 9133i. Thus far the only ideas we've had have involved trying to figure out if you can send back something from the caller ID to change it on the phone, or maybe I could get away with using SMS to send a message and that might be enough for the agents. Any thoughts or suggestions are much appreciated. Thanks, Frank Webb ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 / SIP tftp configs
Matt Schulte wrote: 1) You have to do a factory reset, or wipe out the line config. 2) By default it dials ext 8500 I believe. 3) You *should* be able to change _name, I can't remember the effect that has since you already have authname in. Matt -Original Message- From: Asterisk User Group [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 24, 2005 11:45 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 / SIP tftp configs I have three questions about my 7960 phone that I can't discern from the docs/wiki. 1st - If I change the SIPxx.cnf file to change registrations it sets up new lines as expected. If I delete a line it doesn't get removed when I reboot the phone. I have to go to the phone, unlock it, and reset the SIP parameters. How do I make it forget what it has programmed and listen only to the download? In the SIPphone mac.cnf file put the value UNPROVISIONED into each lineX variable which you want removed. 2nd - Has anyone figured out how to get the Message button to launch a dial to VoicemailMain? Just set the messages_uri: parameter to be the lead number for your voicemail server. 3rd - How do I display on the LCD an alias to the registered line? line1_name: 2000 line1_authname: 2000 line1_password: ** I think you want the lineX_shortname parameter. The doc seems to suggest that line1_name is what it registers with and line1_authname is what it uses if challenged during the authentication. This doesn't make any sense to me. I am looking for the line to be 2000 but the display to say Home or Business, etc. Thanks, dbc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GXP 2000 Firmware 1.0.1.2
Greetings all Grandstream released a new firmware and it seems like the speaker phone problem has been fixed. However we updated to firmware 1.0.1.12 to fix the echo problem but found other problems were now created. The worst of these new problems is that the whole phone starts degrading, the volume starts getting lower and lower. The ringing starts fading and the calls start stuttering. The only way this can be fixed is by rebooting the phone. We were able to replicate this problem in all phones while some Polycoms we have do not suffer from this problem. Again, this problem happened AFTER we upgraded to the new firmware. Has anyone seen this? Jesus Mogollon Global IP Systems, LLC http://www.globalipsystems.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe Marked user?
On Aug 24, 2005, at 11:21 AM, Doug Lytle wrote:[EMAIL PROTECTED] wrote: Hello,But does not go into how to mark a user. voip-info archives, and google didn't lead me to any clue, anddigging to app_meetme.c wasn't fruitful.Anyone have an example on how they marked a user in their dialplan? Create an extension that the user to be marked knows about, maybe even have it authenticate, mark the user and drop them into the conference.DougIf the Marked user isn't the first to enter the channel, then how does the MeetMe app know to put all otherusers on hold until Marked user arrives? This is still unclear to me.ThanksNiles Ingalls___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] realtime sip channel configuration - insecure option
Billy wrote: `insecure` varchar(4) default NULL, This can be changed. I just read the chan_sip.c code and the following values are acceptable: very yes true basically anything with true/false value port invite port,invite invite,port The varchar(4) was originally intended for: very, yes or no -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF not working
Everywhere it is RFC2833 including in SIP phone, Asterisk's sip.conf. DTMF work only from the phone that is hooked with asterisk box. Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Wed, 24 Aug 2005 12:04:04 -0400 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] DTMF not working Innocent Evil wrote: I am having same problem .. DTMF is not working from a SIP phone while sending to Asterisk cmd VoiceMailMain. Have you set DTMF to out of band RFC2833? In band won't work. At least in my version of HEAD John Novack Would you please explain this line !941+1336/100,!0/100, /* 0 */ what value is what and how it affect on DTMF tone generation. Thanks, I had a similar problem that seems to be caused by the DTMF tone lengths being to short. Try this: Asterisk generates DTMF tones in do_senddigit() in the file channel.c. The tones are defined in a const char array called dtmf_tones[]. Each DTMF tone is a string that looks something like: !941+1336/100,!0/100, /* 0 */ The part that reads !941+1336/100 is the part that you want. Change the 100 to something bigger and recompile. You will have to do that for every tone. I'm using 400 right now, and it seems to be working. I hope that helps. Rob Peter Osborne wrote: Hi all, I just upgraded from Asterisk 1.0RC1 to Asterisk 1.0.7 and our dtmf no longer works with external phone systems. I have a Wildcard TDM400P with 4 FXO's? (it connects to analog lines). No changes were made to the config files. Here's my config: /etc/zaptel.conf fxsks=1-4 loadzone = us defaultzone=us /etc/asterisk/zapata.conf [channels] usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=yes rxgain=2.0 txgain=2.0 callgroup=1 pickupgroup=1 musiconhold=default context=incoming group=1 signalling=fxs_ks echocancel=64 echocancelwhenbridged=yes relaxdtmf=yes channel = 1-3 [pete_desk] ;Pete's Desk phone (Polycom IP 300) type=friend username=pete_desk secret=pass context=longdistance callerid=Pete 601 host=dynamic mailbox=601 dtmfmode=inband disallow=all allow=ulaw allow=alaw Thanks, Pete ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Robert Tarte Pacific CodeWorks P.O. Box 29050 San Francisco, CA 94129 (p) 831-426-7582 (f) 831-426-7584 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel Not Sending Tones
Hello everyone, I had an asterisk box which was
working great bt now for some reason I cannot dial out
on any of my outside lines. I am using a TDM card
with 4 FXO ports.
System:
Debian Sarge, 2.6.8-2-386
Compaq Proliant ML370 G2 Server
Polycom IP500 Phones
dtmfmode=rfc2833
I initially set up my system by comiling the latest
cvs zaptel drivers and then the lastest cvs asterisk.
Configured my system and every thing seemed to worked
great. But somehow I changed something or my TDM card
is toasted because now when I dial an outside line I
always receive the telecom standard recording that my
call could not be completed try again
I am using the following extensions.conf line to dial
an outside line:
exten = _9X.,1,Dial,Zap/g1/1-4/${EXTEN:1}
When I dial a number such as 9555 on my phone
asterisk CLI shows this:
-- Executing Dial(SIP/202-115a, Zap/g1/1
4/555) in new stack
-- Called g1/1-4/555
-- Zap/1-1 answered SIP/202-115a
Then I hear afer about 7 seconds the operator message.
I have listened in with an analog handset to what is
happening on the line when asterisk runs this command
is sending tones... but they seem to be not spaced
correctly.
- I tried switching from koolstart signaling to
loopstart no change.
- I tried installing the Sarge zaptel package of
drivers and then recompiling asterisk. No change.
- I tried recompiling latest cvs of zaptel and
asterisk, no change.
I have the following setup in my zaptel.conf
loadzone = us
defaultzone=us
fxsks=1-4 (switched this to fxsls for loopstart
testing no change)
zapata.conf
[trunkgroups]
[channels]
context=default
switchtype=national
signalling=fxo_ls
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=no
restrictcid=yes
usecallingpres=yes
callwaitingcallerid=no
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
group=1
context=default
channel = 1-4
To conclude, I have tested this on different physical
phone lines, different extensions, different channels
on the tdm card and the same result happens. Tested
the phone line without asterisk... no problem lines
work fine?
Any help ?
Thanks all
Kenny
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[Asterisk-Users] [Asterisk-Dev] SIP Benchmarking / Stress Testing
Anyone have a good tool(s) to use for simulating a bunch of calls? Benchmarking or stress testing? I only need SIP protocol, and do appreciate any replies...I realize I could google it, but I am looking for opinions as well. Sherwood McGowan ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Asterisk-Dev] SIP Benchmarking / Stress Testing
Sherwood McGowan wrote: Anyone have a good tool(s) to use for simulating a bunch of calls? Benchmarking or stress testing? I only need SIP protocol, and do appreciate any replies...I realize I could google it, but I am looking for opinions as well. There is SIPp: http://sipp.sourceforge.net/ Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PhoneCALL version 1.0 Administrative Manual - Released
Greetings Everyone! The version 1.0 of the PhoneCALL Administrative Manual has been released. It is more of an outline of the features and interface, and we'll be adding lots of more detailed information in the manual over the next few days/weeks. Of course, we'd love to get your input on the manual and areas we need to clarify or even some new sections in the manual that would help explain PhoneCALL and how it works. You can find the PDF version of the manual in the Downloads, or you can view the HTML version here: http://www.vecsector.com/phonecall/demo/manual Enjoy! Dustin Wildes VecSector, LLC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HooDaHek 0.4 Released
HooDaHek, the caller ID and instant messaging notification service for Asterisk boxen, is now updated to version 0.4. Information/download here: http://www.nathanpralle.com/software/hoodahek.html Changes: - Changed the AIM bot to use Net::OSCAR instead of Net::AIMTOC since AOL managed to break TOC in some way. Tired of such shenanigans, so switched. Reminder to self: Call Apple. - Implemented HiRes timing for the Bot so it pops up CallerID information tons faster than it used to -- often by the first ring. Enjoy. Nathan -- - Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: {Scanned} Re: [Asterisk-Users] Caller ID ?
Stijn Jonker wrote:
Hello Tom,
On 26-Aug-2005 7:50, Tom wrote:
Most of the time i can find answers to my questions on the wiki, google,
or searching the list now i am stuck .
I have a small * box at my house running 1.0.9 stable and a devlite kit.
Every thing is awesome VM, IVR, Echo canceling, and Meetme are all
working great.
Nice isn't it?
But on Incoming caller id i need to add a 9 as a prefix to make it
easier to return call from my cordless phone (cheap vtech phone). I have
tried to search the list and also google but i think i am searching of
the wrong thing. If i could get a kick in the right direction that would
be great.
This is what I came up with: (Watch out for linewraps on the second line.)
; Incoming on normal line
; Incoming on normal line
exten = ${EDN_MAIN},1,LookupCIDName(${CALLERIDNUM})
exten = ${EDN_MAIN},2,GotoIf($[$[${CALLERIDNUM} = ] |
$[${CALLERIDNUM} = CID withheld]]?5:3)
exten = ${EDN_MAIN},3,SetCIDNum(9${CALLERIDNUM})
exten = ${EDN_MAIN},4,SetVar(__NETWORK=KPN-Prive)
exten = ${EDN_MAIN},5,Goto(int-dest,${EDN_MAIN},1)
Stijn
Stijn,
Thanks based on my reading on the wiki i thought the the cmd SetCIDNum()
was only for forcing Caller id on a PRI.. :-[ once again thanks for the kick
BTW:: this list rocks :-) so much good info
Tom
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RE: [Asterisk-Users] IAX2 Softphone Quality Network Cards
I have had similar experience with an Intel NIC that had DELL's name on it vs a 3COM 3C905b. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Logan Sent: Friday, August 26, 2005 10:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX2 Softphone Quality Network Cards I haven't had similiar experience, but in several threads about sound quality people have talked about Network cards being the culprit. In particular, a few people have commented all sorts of problems on onboard NIC's, since they tend to be of lesser quality than stand-alone NICS. On 8/26/05, Adam Robins [EMAIL PROTECTED] wrote: We are in the process of an Asterisk call center deployment using IAX2 G711 ulaw softphones. Outbound sound quality is terrible. This week we rebuilt the entire LAN with Cisco 2950-EI switches and have employed QoS on the switches and router. Still sounds terrible. What we are now finding is that the network card in the PC may be the key to the problem. A Dell Optiplex P4 2.4GHz 512MB machine with an onboard Intel NIC is bad, while an older Dell Dimension P3 864MHz 128MB machine with onboard 3COM sounds good. Has anyone out there had a similar experience? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk: Unable to read password.
Hi Pat,
I would check the DTMF settings on your phone - I had a similar problem
until I switched to RFC from Inband.
Regards,
--
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Business Systems Analyst
District of North Vancouver
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On Aug 26, 2005, at 4:56 AM, pat newham wrote:
Hello,
I am using asterisk as voicemail for my sip proxy.
When a user (1234)dials , the call is forwarded to
asterisk. However I receive the following error:
--Executing VoiceMailMain(SIP/1234-9afc, 1234) in
new stack
--Playing 'vm-password' (language 'en')
[WARNING]: app_voicemail.c:3359 vm-execmain: Unable to
read password
==Spawn extension (default, , 1) exited non-zero
on 'SIP/1234-9afc'
My configs are as follows:
;sip.conf
[1234]
type=friend
host=dynamic
context=default
mailbox=1234
;extensions.conf
[default]
exten=1234, 1, Voicemail(u${EXTEN})
exten=1234, 2, Hangup
exten=, 1, VoicemailMain(${CALLERIDNUM})
;voicemail.conf
1234=1234, P, [EMAIL PROTECTED]
Please advise if possible as i have looked through the
asterisk mail archives but cannot see what would be
wrong with the configuration.
many thanks.
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[Asterisk-Users] ChanIsAvail for IAX not working again/still? AKA Redundant IAX connections not working
Hi -
I'm running CVS-HEAD from 2005-08-11 20:17:17 UTC, and I'm trying to
set up some redundancy on IAX connections between locations. I have
two IAX peers set up that work correctly by themselves: ast551-out
and ast551-out-backup:
[ast551-out]
type=peer
secret=secret
username=ast551
host=X.X.X.X
qualify=1000
disallow=all
allow=gsm
allow=ulaw
trunk=no
tos=0x04
[ast551-out-backup]
type=peer
secret=secret
username=ast551-backup
host=Y.Y.Y.Y
qualify=1000
disallow=all
allow=gsm
allow=ulaw
trunk=no
tos=0x04
If one does become unavailable, I'd like the other to be used. I
tried to set that up like this:
exten = 145,1,ChanIsAvail(${IAX2/iax-in:[EMAIL PROTECTED]/$
{EXTEN})
exten = 145,2,Dial(${IAX2/iax-in:[EMAIL PROTECTED]/${EXTEN},
20,t)
exten = 145,102,Dial(${IAX2/iax-in:[EMAIL PROTECTED]/${EXTEN},20,t)
What is happening is that all calls are going out through ast551-out-
backup, even when I physically disable the connection. The console
shows this:
-- Hungup 'IAX2/ast551-out-backup-2'
-- Executing Dial(SIP/68-1c7a, IAX2/iax-in:[EMAIL PROTECTED]
backup/145|20|t) in new stack
-- Called iax-in:[EMAIL PROTECTED]/145
-- IAX2/ast551-out-backup-7 is circuit-busy
Aug 26 12:11:56 NOTICE[14283]: chan_iax2.c:2736 auto_congest: Auto-
congesting call due to slow response
-- Hungup 'IAX2/ast551-out-backup-7'
== Everyone is busy/congested at this time (1:0/1/0)
Doing an iax2 show peers shows ast551-out-backup to be offline:
ast33*CLI iax2 show peers
Name/UsernameHost Mask Port
Status
astnh-out/ast55 Z.Z.Z.Z (S) 255.255.255.255 4569
Unmonitored
ast551-out-back Y.Y.Y.Y (S) 255.255.255.255 4569
UNREACHABLE
ast551-out/ast5 X.X.X.X (S) 255.255.255.255 4569 OK (25
ms)
3 iax2 peers [1 online, 1 offline, 1 unmonitored]
Have I bumbled a configuration, or is my method incorrect? Or is
there a bug? Should ChanIsAvail report that ast551-out-backup is
unavailable if it fails to qualify?
Thanks,
Noah
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[Asterisk-Users] RE: Voicetronix openline4 quality
I am looking at alternatives to the Digium TDM04B. The only one I can find is the Voicetronix openline4 but I cannot find a lot of information on it. Does anyone have any experience with it on Asterisk that they can compare to a Digium TDM04B. I am particularly interested in the built in hardware echo canceller and how well it works as opposed to software echo cancellation with a Digium card. Does anyone have it working in a production PBX environment? Does it work reliably on Asterisk. Is the a quality product? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared
So bottom line please. Have we decided that it is STILL correct to set RX/TX gain for 14800 with ztmonitor quantitative using a telco 1004hz 0dbm test phone number? If not, what should we set it to with ztmonitor. -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Thursday, August 25, 2005 8:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared I'll do my comments in line and hope I don't offend. Rich Adamson wrote: First off, thank you *very* much for this unbelievably informative post! I've got it saved away now along with Kris Boutilier's adjusting rxgain/txgain post. On Wednesday 24 August 2005 17:14, Bruce Ferrell wrote: At the point where the phone line get's to your demarc the is supposed to ba a -2 to 3db reference point, sometimes called a -2 or -3 test level point (TLP). So that milliwatt tone at that point should read in the range of -2 to -3 dbm. If I read the above words exactly as written, the above is not true. Maybe there was a different intent that I'm missing, or, maybe words left out? I'm a lousy typist :) I'm reading the words to say if I put a transmission test set on the cable pair just before the pair leaves the central office, the reading should be in the -2 to -3 dbm range. If that is what you meant, then its incorrect. Even the old analog step-by-step switch specs called for no more then .5db loss from the milliwatt generator to the cable pair (CO distribution frame). If you mean placing a transmission test set at the customer's demarc (at the customer's site), the -2 to -3 db is still incorrect for analog pstn circuits. That level _will be_ the 0db generator tone minus the cable loss from the CO to the customer's demarc. That cable loss is 100% predictable if you know the length and gauge of the copper wires between the central office and the customer's site. (That is exactly how the engineering spec is set for the less technical telephone installers to measure after installing a new pstn facility to a customer site.) at the last point leaving the CO, the tone level should be a nominal 0dbm. By the time it get's to the customer demarc, -2 to -3 dbm. The loops are suppposed to be engineered that way. On a brand spanky new loop, yes 100% predictable. Over time, all sorts of oddities (corrosion, half taps, loading coils, and just general funkieness) are introduced in the real world. The -2 to -3 db is not correct for analog circuits. Copper wires have a loss that is directly related to the length of the cable. (I don't have the chart right here, but a 7,000 foot cable pair will have lets say 6db of loss and a 3,000 foot pair will be a 3db loss. You can't engineer something into a copper pair to compensate for that loss.) The only thing that I can think of that you might be talking about is using an old analog carrier system on a copper pair. If that's what you're thinking, then yes -2 to -3 db is very reasonable. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] fedora core 3 kernel source - could someonethrowthe dog a bone!
What was the issue with zaptel and 2.6.12? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer Sent: Thursday, August 25, 2005 1:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] fedora core 3 kernel source - could someonethrowthe dog a bone! I found that only the kernel is installed. I'd avoid 2.6.12 for now as I had problem with the zaptel driver and stay with 2.6.9. Regards Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: 24 August 2005 22:33 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] fedora core 3 kernel source - could someone throwthe dog a bone! This could be a duplicate post, sent it originally 4 hours ago, it never showed up! I know this is a question with an obvious answer to some, but I am not one of them. Installed FC3, but this time I decide to update since my ISOs are a bit old, so typical yum update Downloaded the FC3 SRPM for my kernel 2.6.12 Installed the SRPM package Ran rpmbuild bp target=i686 kernel-2.6.spec Tried to build zaptel error; You do not appear to have the sources for the 2.6.12-1.1372_FC3smp kernel installed. So I assume that either a) I did not build the correct source for the smp kernel, or b) I am missing a symbolic link to the kernel source. No help from the FC3 release notes, no help from a Google. So, if you dont mind, throw me the bone ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom Phone advise
I would like to know if any body is using the Polycom Soundstation IP 4000 SIP conference phone with Asterisk. I am thinking of purchasing one. Kurt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Phone advise
I have one and it is absolutely awesome. Works great and the quality of Polycom conference phones is excellent regardless of protocol. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of kurt x Sent: Friday, August 26, 2005 9:50 AM To: Asterisk Subject: [Asterisk-Users] Polycom Phone advise I would like to know if any body is using the Polycom Soundstation IP 4000 SIP conference phone with Asterisk. I am thinking of purchasing one. Kurt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AstTAPI Config File Location
Hello List!!! Does anyone know where AstTAPI stores it's configuration information? We have a domain and our users are 'domain users'. When they change the TAPI registration information in AstTAPI it does not stick. Certainly, I don't have to give every admin permission on their box for this? Thanks, Bill ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Attached Voicemail does not play mac/linux
Hi, I noticed the .WAV file for voicemails is what gets e-mailed to people when someone leaves a voicemail. I also noticed today that I can not play the .WAV files on my macintosh or linux machines. I *can* play the .WAV files on my Windows machines. I can play the .wav files on either machine. Can someone explain what's different about the .WAV files and how do I get them to play on Macintosh and Windows machines? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] When 486 ATA crashes, asterisk does not disconnect the call
Hi, On several occasions one or more of our grandstream Handy tone 486 ATA would crash. If for some reason that ATA is not rebooted immediately, asterisk would not disconnect the call, even though the party on the other end of the call have already hung up the call. The call would continue via my asterisk server and my sip termination provider indefinitely until I either reboot the ATA device or restart asterisk. It even ignores the timeout setting for the call. Can anyone explain why that would happen and how I can resolve that problem. Thanks Joel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Polycom Phone advise
Hi Kurt - I would like to know if any body is using the Polycom Soundstation IP 4000 SIP conference phone with Asterisk. I am thinking of purchasing one. Yes, we have one, and we have the add-on pod mics for it, too. The setup works well, for the most part. The mics aren't quite as sensitive as I'd like them to be. I'm sure it has to do with keeping out backround noise, but if people do not speak loudly and clearly, the mics won't pick them up. The speaker is mostly good, but a bit muddy. That probably has to do with the fact that our asterisk ulaw calls are only 8-bit, though. The screen is also VERY small, but it is bright. On the whole, it is definitely better than trying to use an IP500 for the same task, especially for a large space with a lot of people. - Noah ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 Softphone Quality Network Cards
We are using Plantronics H51N headset top with DA55 USB adapter which has DSP built-in. Terrible means garbled, unintelligible, underwater-sounding. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: Friday, August 26, 2005 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX2 Softphone Quality Network Cards Hi! We are in the process of an Asterisk call center deployment using IAX2 G711 ulaw softphones. Outbound sound quality is terrible. Have you tried a different sound card and/or a USB handset (which includes an external sound card)? And what exactly do you mean with terrible sound? Philipp ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fedora Core 4 x86_64
I am about to build a Dual Opteron Asterisk box as our soon to be production server. Is Core 4 supported or should I stay with Core 3? There was a recent post about an issue with the latest Core 3 Kernel and zaptel. I had the same experience, but just rolled back to the previous version of the Kernel on Core 3 on our evaluation server. Thanks in advance ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attached Voicemail does not play mac/linux
Try format=wav|gsm instead of format=wav49|gsm in your voicemail.conf Be advised, however, that the attached files will be considerably larger - we made this change to increase the volume of attached messages, and can live with the increased file size. I use a Mac, and the files play just fine. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Aug 26, 2005, at 10:09 AM, Matt wrote: Hi, I noticed the .WAV file for voicemails is what gets e-mailed to people when someone leaves a voicemail. I also noticed today that I can not play the .WAV files on my macintosh or linux machines. I *can* play the .WAV files on my Windows machines. I can play the .wav files on either machine. Can someone explain what's different about the .WAV files and how do I get them to play on Macintosh and Windows machines? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ignorepat not working - what might I have done?
Hey, all. I have the following, and
ignorepat = 9
; Testing - access to telco1/FXO
; XXX
exten = _9.,1,Dial(SIP/outboundfxo/${EXTEN:1},20)
exten = _9.,2,Hangup
Unfortunately, once I hit 9 on a connected phone, I do *not* get a dial
tone back.
Can someone suggest what I might have done wrong?
Thanks!
--
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I am a brother of jackals, and a companion of ostriches. (Job 30 : 29)
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[Asterisk-Users] Ztmonitor values when zap channel is onhook
Hello, In my quest to figure out the source of the random echo on our shiny new asterisk install, I have been using ztmonitor on the TDM400p channels for the good part of today. I have 2 TDM400p cards with 8 FXO modules and 6 pots lines connected to them (last 2 channels are unused but configured in zaptel). Even when the lines are onhook, the Tx values settle down to 0 but the Rx values still jump up and down. For some lines the values vary around 440, for others around 250 and for one of the unused FXO ports, its around 97 all the time. With txgain/rxgain set to 0.0, call volumes were considerably low. Hence, I have txgain set to 1.0 and rxgain set to 6.0. I was wondering if the ztmonitor Rx: values were normal for what they show, or are they too supposed to settle down to 0. If this is not normal, does this indicate any problems with the pots lines? -- VaibhaV ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fedora Core 4 x86_64
Take it from someone who owns 25 of them. Stay away from FC anything. Use CentOS 4 its better more stable and has true multi-treading as FC doesn't thread anything.. ..o---o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Supporter Sent: Friday, August 26, 2005 1:16 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Fedora Core 4 x86_64 I am about to build a Dual Opteron Asterisk box as our soon to be production server. Is Core 4 supported or should I stay with Core 3? There was a recent post about an issue with the latest Core 3 Kernel and zaptel. I had the same experience, but just rolled back to the previous version of the Kernel on Core 3 on our evaluation server. Thanks in advance ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ignorepat not working - what might I have done?
Mason Loring Bliss wrote:
Hey, all. I have the following, and
ignorepat = 9
; Testing - access to telco1/FXO
; XXX
exten = _9.,1,Dial(SIP/outboundfxo/${EXTEN:1},20)
exten = _9.,2,Hangup
Unfortunately, once I hit 9 on a connected phone, I do *not* get a dial
tone back.
ignorepat does not work for SIP since the dialtone is coming from the
SIP device, not from Asterisk.
You would need to set the phone up to continue dialtone after dialing 9.
Not all phones support that.
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Re: [Asterisk-Users] Ztmonitor values when zap channel is onhook
VaibhaV Sharma wrote: Hello, In my quest to figure out the source of the random echo on our shiny new asterisk install, I have been using ztmonitor on the TDM400p channels for the good part of today. I have 2 TDM400p cards with 8 FXO modules and 6 pots lines connected to them (last 2 channels are unused but configured in zaptel). Even when the lines are onhook, the Tx values settle down to 0 but the Rx values still jump up and down. For some lines the values vary around 440, for others around 250 and for one of the unused FXO ports, its around 97 all the time. With txgain/rxgain set to 0.0, call volumes were considerably low. Hence, I have txgain set to 1.0 and rxgain set to 6.0. why not txgain=-6 to start out with? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: asterisk in Taiwan
On 8/25/05, Lance Grover [EMAIL PROTECTED] wrote: I have now tried the lattest zaptel drivers for the 4 port tdm card (wctdm) and it still cuts off after the 10 - 15 seconds. Any Ideas? I now found the issue, in the extensions.conf file I had the variable TRUNK=Zap/g2c to check the channel before dialing but by taking off the 'c' on the end it fixed this issue that I was having. Just FYI. -- Thanks, Lance Grover ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: SV: [Asterisk-Users] Cisco and protocol application invalid
Thanks for the suggestions everyone! The thing is, when I run tcpdump, this phone never really requests anything as far as I can see. The IP-address serving is not handled by the Asterisk box, which is on its dedicated IP-address, but by a consumer type SMC Barricade 4-port router. There's no option to define boot tftp on this device. I'm thinking of getting myself a console cable for the phone, and see if I'm able to log in and set the variables using that. Don't know if these cables can be purchased in stores or if they'll have to be hand made.. My biggest problem is that I simply do not know which IP-address the phone is searching for - if its searching for any IP address at all. Thanks again, have a great weekend. Bjorn -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Tom Rymes Sendt: 16. august 2005 16:47 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: [Asterisk-Users] Cisco and protocol application invalid On Aug 15, 2005, at 11:28 AM, Bjørn Ove Kristiansen wrote: Hello! The issue is simply that I don't know which IP address the phone tries to connect to. I am not very familiar with dhcpd (never put it up by hand), so I'm not sure how the below would help me, but from what I can tell, I still need information on which IP-address the phone is trying to find its tftp on, right? Bjorn [snip] Michiel van Baak wrote: I have put this in my dhcpd.conf to make sure my cisco phones connect to my TFTP server: server-name 192.168.2.1; I'd be surprised if that worked... the server name is for.. um.. the name of the server :) Try: option tftp-boot-server code 150 = ip-address; option tftp-boot-server 192.168.44.3; [snip] Bjorn, The phone will use whatever TFTP server your DHCP server tells it to use. That is what the above line option tftp-boot-server is supposed to do, it tells your DHCP server what address to give your phone. If your DHCP server does not give the phone an address, and you have not specified one manually through the phone's settings interface, then it will (in my experience) default to the same address as your DHCP server. What are you using for a DHCP server? Can you enter the settings interface of the phone if you leave the ethernet disconnected? (If so, manually specify the TFTP server address.) Tom___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.10/73 - Release Date: 15.08.2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.15/82 - Release Date: 25.08.2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Softphone Quality Network Cards
tourn of AGC , and mybe use GSM. for usb device that use iax2 prtocol there are this one that have nice http://www.gedameurope.com/us/002servizi_e_prodotti%5Bus%5D.htm this usb device doe not need external sound card. Philipp von Klitzing wrote: Hi! We are in the process of an Asterisk call center deployment using IAX2 G711 ulaw softphones. Outbound sound quality is terrible. Have you tried a different sound card and/or a USB handset (which includes an external sound card)? And what exactly do you mean with terrible sound? Philipp ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Informazione NOD32 1.1202 (20050825) __ Questo messaggio è stato controllato dal Sistema Antivirus NOD32 http://www.nod32.it -- Cheers Andrea Andrea Cristofanini Gedam Europe S.r.l. Gedam Advanced Communication LTD mobile : +39 3291871756 office : +39 011 5694900 MSN : [EMAIL PROTECTED] www.gedameurope.com www.asterisknews.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicetronix
Anybody using voicetronix cards? The 12 ports for example? What has been your experience and how many cards can be put into one server? Do they have the same IRQ problems as Digium ones? AK ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 911 Notices
With the deadline coming up for sending notices to customers, I found it curious that out of 4-5 different providers I use, to date only one of them has contacted me. The rest don't even have anything on their website that I could find. Junction Networks was the only one that actually sent me a letter and also have everything right on the first page when you login to their system. A week or so ago I remember reading an article where the CEO from one of my vendors was complaining that they wouldn't have enough time to get all of their customers to respond in time. I thought that was pretty funny given that they don't seem to even be contacting anyone yet. There isn't even anything on their website except a statement that they do not plan to support 911 anytime soon. Am I missing something here? Is the FCC going to be extending deadlines and that's why the apparent lack of action on this issue? Just curious. I thought I would have started receiving letters a long time ago. Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 Notices
An extension of 30 days has been granted. Just like the HDTV broadcast requirement deadlines the FCC cooked up I'd predict there will be a few more extensions before the fight is over. http://tinyurl.com/a8tj8 -- Reuters article Am I missing something here? Is the FCC going to be extending deadlines and that's why the apparent lack of action on this issue? Just curious. I thought I would have started receiving letters a long time ago. Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ztmonitor values when zap channel is onhook
On Fri, 2005-08-26 at 12:37 -0500, Eric Wieling aka ManxPower wrote: VaibhaV Sharma wrote: Hello, I have 2 TDM400p cards with 8 FXO modules and 6 pots lines connected to them (last 2 channels are unused but configured in zaptel). Even when the lines are onhook, the Tx values settle down to 0 but the Rx values still jump up and down. For some lines the values vary around 440, for others around 250 and for one of the unused FXO ports, its around 97 all the time. With txgain/rxgain set to 0.0, call volumes were considerably low. Hence, I have txgain set to 1.0 and rxgain set to 6.0. why not txgain=-6 to start out with? With txgain set to 0.0, people on the other end complained about low volume from our side hence I started out with a positive value. Are the ztmonitor Rx: values of around 90 - 400 normal even when the zap channel is Onhook or do I have some problem on the pots lines that I should fix? Thanks, -- VaibhaV ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] French national telco 1004hz test phone number ?
Hello Asterisk friends, Does somebody know few french phone numbers to do telco 1004Hz 0dBm signal tests phone ? Thanks in advance. Best Regards, Francois BERGERET, Happy French Asterisk user :-) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PCI 2.3
Title: PCI 2.3 Hello All, Anyone know if this is backwards compatible with 2.2? Here is the spec from the Mobo I am looking at. Five 32-bit v2.3 Master PCI bus slots (support 3.3V/5V PCI bus interface). Thanks! Wley ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS HEAD HDLC Abort on a TE405P PRI
I just upgraded my asterisk install from january 2005 CVS HEAD to current CVS HEAD : * zaptel * libpri * asterisk My asterisk have one TE405P with one span (the clock source) pluged into a telco PRI E1, a second span is a PRI E1 to another PBX and the third one is a T1 to a rhino channel bank. Everything went smoothly except that the two PRI span restart every ten seconds after some HDLC abort errors : Aug 26 20:04:15 NOTICE[20571] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 Aug 26 20:04:15 NOTICE[20571] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 Aug 26 20:04:30 NOTICE[20571] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Aug 26 20:04:30 NOTICE[20571] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 Aug 26 20:04:55 NOTICE[20571] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Aug 26 20:05:10 NOTICE[20571] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Aug 26 20:05:10 NOTICE[20571] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 Aug 26 20:05:55 NOTICE[20571] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 ... I can place and receive calls but they are randomly droped and full of clicks After reverting only the zaptel driver everythings works normaly again. One of the reason that lead me to do the upgrade is to test the new echo canceler. But unfortunatly, I can't test it as it's in the zaptel modules. Any ideas. cyril ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voice modification
Anyone played with the possibility of modifying how a voice sounds on asterisk? Eg make an outbound call from asterisk but by pressing *1 your voice goes into a higher pitch etc? Just a thought, Cheers, Dean ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared
I'm not the OP, but I had a similar problem, in my case fxotune ran successfully for just one out of 3x FXO modules, but the coefficients were all 0's. My kernel is 2.6.11 on CentOS 4.1. So I'm curious if 2.6 kernel (instead of 2.4) has any input in this whole echo issue, not just fxotune. Yesterday I switched to KB1 echo canceller, it is by far the best. But today I had a similar experience to Eric Rees's Strange Echo post. After transfering to another internal line, echo starts. My theory is that after transfer some characteristics of the internal connection change, especially the Tx voice (the person talking on our side changes). So if the echo canceller is too committed to the voice of the first person answering the line (the operator), that would be quite expected. I don't know how KB1 or other echo cancellers work, but if I'm right, it would be better if echo canceller readjusted itself after transfer. Sorry if that's plain wrong. Can somebody comment please? I'm really interested in all posts in this thread and others or documents on echo. Btw, thanks Eric Wieling for the Cisco link. That article is an excellent read. Readers should be a little carefull with it however as there can be additional sources not mentioned in the article. Others that have more capability to read code then I might want to comment on the following to help ensure we're all running with the same reasonable understanding. Relative to asterisk's canceler and based on two years of rather heavy experience with asterisk, one can characterize the existing canceler(s) by saying there are two distinct functional pieces: 1. the pstn line pulsing used to preload the canceler, and, 2. the ongoing real-time training. The first function is controlled by 'echotraining=800' (or whatever value including 'yes' might be provided) in zapatal.conf. The second part can actually be heard in most implementations by changing echotraining=no and listen to an actual call. Typically, it takes about ten seconds or so for the training to occur. (The actual time varies depending upon how good/bad the end-to-end circuit happens to be. Is it practical to 'assume' that in your case mentioned above that #1 is not going to occur again (since I assume when you say 'line' you are referring to an outside pstn line), and, #2 is in a mode of fine-tuning the training when in fact you'd really like it to start the coarse-training from scratch? Relative to the fxotune app, it would appear the app is specific to the v2.4 kernels (/dev/zap*), which the v2.6 kernels don't use (but rather the udev equivalent). (When I had * running on a v2.4 kernel, the output from fxotune never deviated from all zero's. So I'm assuming the default chipset values were already tweaked by the chipset manufacturer to US telco lines. If that is true, then running fxotune in the US has very little value.) The KB1 canceler _does_ work just fine in the v2.6 kernels and I'm in favor of moving it to the default for future Head and Stable releases as soon as practical. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared
Bottom line... ztmonitor can be used to 'assist' in setting some starting values, but the further your asterisk box is from the central office, the more likely the gain values will have to be adjusted lower then what you want, and may very well appear off-scale with ztmonitor. Given the curent code and issues, using your ears instead of ztmonitor will lead to better results, period. (Before lots of people jump on this and say it does work, please reread the further you are from the CO words again. Yes, ztmonitor can be used with low-loss pstn loops; no, it will not provide anything close to an optimal circuit for higher-loss loops.) So bottom line please. Have we decided that it is STILL correct to set RX/TX gain for 14800 with ztmonitor quantitative using a telco 1004hz 0dbm test phone number? If not, what should we set it to with ztmonitor. -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Thursday, August 25, 2005 8:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared I'll do my comments in line and hope I don't offend. Rich Adamson wrote: First off, thank you *very* much for this unbelievably informative post! I've got it saved away now along with Kris Boutilier's adjusting rxgain/txgain post. On Wednesday 24 August 2005 17:14, Bruce Ferrell wrote: At the point where the phone line get's to your demarc the is supposed to ba a -2 to 3db reference point, sometimes called a -2 or -3 test level point (TLP). So that milliwatt tone at that point should read in the range of -2 to -3 dbm. If I read the above words exactly as written, the above is not true. Maybe there was a different intent that I'm missing, or, maybe words left out? I'm a lousy typist :) I'm reading the words to say if I put a transmission test set on the cable pair just before the pair leaves the central office, the reading should be in the -2 to -3 dbm range. If that is what you meant, then its incorrect. Even the old analog step-by-step switch specs called for no more then .5db loss from the milliwatt generator to the cable pair (CO distribution frame). If you mean placing a transmission test set at the customer's demarc (at the customer's site), the -2 to -3 db is still incorrect for analog pstn circuits. That level _will be_ the 0db generator tone minus the cable loss from the CO to the customer's demarc. That cable loss is 100% predictable if you know the length and gauge of the copper wires between the central office and the customer's site. (That is exactly how the engineering spec is set for the less technical telephone installers to measure after installing a new pstn facility to a customer site.) at the last point leaving the CO, the tone level should be a nominal 0dbm. By the time it get's to the customer demarc, -2 to -3 dbm. The loops are suppposed to be engineered that way. On a brand spanky new loop, yes 100% predictable. Over time, all sorts of oddities (corrosion, half taps, loading coils, and just general funkieness) are introduced in the real world. The -2 to -3 db is not correct for analog circuits. Copper wires have a loss that is directly related to the length of the cable. (I don't have the chart right here, but a 7,000 foot cable pair will have lets say 6db of loss and a 3,000 foot pair will be a 3db loss. You can't engineer something into a copper pair to compensate for that loss.) The only thing that I can think of that you might be talking about is using an old analog carrier system on a copper pair. If that's what you're thinking, then yes -2 to -3 db is very reasonable. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr_odbc in CVS-HEAD gives connect error on reload
If I fire up asterisk it connects to my MSSQL server via ODBC just fine. However, if I issue a reload it unloads the ODBC.. then loads it again and I get an error... and keep getting it until I On a fresh start: Aug 26 15:43:14 WARNING[13818] cdr_odbc.c: cdr_odbc: table not specified. Assuming cdr Aug 26 15:43:14 VERBOSE[13818] logger.c: -- cdr_odbc: dsn is MSSQL-asterisk Aug 26 15:43:14 VERBOSE[13818] logger.c: -- cdr_odbc: username is voip Aug 26 15:43:14 VERBOSE[13818] logger.c: -- cdr_odbc: password is [secret] Aug 26 15:43:14 VERBOSE[13818] logger.c: -- cdr_odbc: table is cdr Aug 26 15:43:14 VERBOSE[13818] logger.c: cdr_odbc: Connected to MSSQL-asterisk On a reload: Aug 26 15:43:57 VERBOSE[13818] logger.c: -- Reloading module 'cdr_odbc.so' (ODBC CDR Backend) Aug 26 15:43:57 VERBOSE[13818] logger.c: cdr_odbc: Disconnecting from MSSQL-asterisk Aug 26 15:43:57 VERBOSE[13818] logger.c: cdr_odbc: free dsn Aug 26 15:43:57 VERBOSE[13818] logger.c: cdr_odbc: free username Aug 26 15:43:57 VERBOSE[13818] logger.c: cdr_odbc: free password Aug 26 15:43:57 VERBOSE[13818] logger.c: cdr_odbc: free table Aug 26 15:43:57 VERBOSE[13818] logger.c: == Unregistered 'ODBC' CDR backend Aug 26 15:43:57 VERBOSE[13818] logger.c: == Parsing '/etc/asterisk/cdr_odbc.conf': Aug 26 15:43:57 VERBOSE[13818] logger.c: == Parsing '/etc/asterisk/cdr_odbc.conf': Found Aug 26 15:43:57 DEBUG[13818] cdr_odbc.c: cdr_odbc: Logging uniqueid Aug 26 15:43:57 DEBUG[13818] cdr_odbc.c: cdr_odbc: Not logging in GMT Aug 26 15:43:57 WARNING[13818] cdr_odbc.c: cdr_odbc: table not specified. Assuming cdr Aug 26 15:43:57 VERBOSE[13818] logger.c: -- cdr_odbc: dsn is MSSQL-asterisk Aug 26 15:43:57 VERBOSE[13818] logger.c: -- cdr_odbc: username is voip Aug 26 15:43:57 VERBOSE[13818] logger.c: -- cdr_odbc: password is [secret] Aug 26 15:43:57 VERBOSE[13818] logger.c: -- cdr_odbc: table is cdr Aug 26 15:43:57 VERBOSE[13818] logger.c: cdr_odbc: Error SQLConnect -2 Aug 26 15:43:57 ERROR[13818] cdr_odbc.c: cdr_odbc: Unable to connect to datasource: MSSQL-asterisk Aug 26 15:43:57 VERBOSE[13818] logger.c: -- cdr_odbc: Unable to connect to datasource: MSSQL-asterisk Another reload and it's fine again (no restart this time): Aug 26 15:44:19 VERBOSE[13818] logger.c: -- Reloading module 'cdr_odbc.so' (ODBC CDR Backend) Aug 26 15:44:19 VERBOSE[13818] logger.c: cdr_odbc: free dsn Aug 26 15:44:19 VERBOSE[13818] logger.c: cdr_odbc: free username Aug 26 15:44:19 VERBOSE[13818] logger.c: cdr_odbc: free password Aug 26 15:44:19 VERBOSE[13818] logger.c: cdr_odbc: free table Aug 26 15:44:19 VERBOSE[13818] logger.c: == Unregistered 'ODBC' CDR backend Aug 26 15:44:19 VERBOSE[13818] logger.c: == Parsing '/etc/asterisk/cdr_odbc.conf': Aug 26 15:44:19 VERBOSE[13818] logger.c: == Parsing '/etc/asterisk/cdr_odbc.conf': Found Aug 26 15:44:19 DEBUG[13818] cdr_odbc.c: cdr_odbc: Logging uniqueid Aug 26 15:44:19 DEBUG[13818] cdr_odbc.c: cdr_odbc: Not logging in GMT Aug 26 15:44:19 WARNING[13818] cdr_odbc.c: cdr_odbc: table not specified. Assuming cdr Aug 26 15:44:19 VERBOSE[13818] logger.c: -- cdr_odbc: dsn is MSSQL-asterisk Aug 26 15:44:19 VERBOSE[13818] logger.c: -- cdr_odbc: username is voip Aug 26 15:44:19 VERBOSE[13818] logger.c: -- cdr_odbc: password is [secret] Aug 26 15:44:19 VERBOSE[13818] logger.c: -- cdr_odbc: table is cdr Aug 26 15:44:19 VERBOSE[13818] logger.c: == Parsing '/etc/asterisk/sip_notify.conf': Aug 26 15:44:19 VERBOSE[13818] logger.c: == Parsing '/etc/asterisk/sip_notify.conf': Not found (No such file or directory) Aug 26 15:44:19 VERBOSE[13818] logger.c: cdr_odbc: Connected to MSSQL-asterisk Anyone have any ideas? my cdr_odbc.conf says: [global] dsn=MSSQL-asterisk username=voip password= loguniqueid=yes and my odbc.ini file is: [MSSQL-asterisk] description = Asterisk ODBC for MSSQL driver = FreeTDS server = xxx.xxx.xxx.xxx port= 1433 database= VoIP tds_version = 8.0 language= us_english Any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Busy number signalling
Andres,
Thanks for the suggestion. I did try it but it is not moving to the
next priority after the Dial command. I also do know for a fact that
it is not actually being answered. On the console I just get:
-- Called g1/123456789
On 8/26/05, Andres [EMAIL PROTECTED] wrote:
Eric Bishop wrote:
Hi all,
Our Asterisk box sends calls outbound via either SIP (through our VoIP
provider) or an E1 PRI (directly connected via a TE410P). When we dial
a number that is engaged via our VoIP provider we get the following on
the Asterisk console (numbers and IP addresses changed to protect the
innocent):
-- Called [EMAIL PROTECTED]
-- Got SIP response 486 Busy here back from 123.123.123.123
-- SIP/sip-outbound-af71 is busy
== Everyone is busy/congested at this time
This is what we want as it then send the call to priority n+101 and we
can handle it any way we want from there. However if the outbound call
is made via the PRI to an enaged number it simply plays an enaged
signal to the caller and never progresses to priority n+101.
Anyone have any suggestions?
You can try checking for the DIALSTATUS variable. ON our PRIs we do
something like:
exten = _1XX,1,Dial(Zap/r1/${EXTEN:1})
exten = _1XX,2,GotoIf($[${DIALSTATUS} = NOANSWER]?8)
exten = _1XX,3,GotoIf($[${DIALSTATUS} = BUSY]?8:4)
exten = _1XX,4,Congestion
exten = _1XX,8,Busy(1)
But if your call is really getting answered then this won't work.
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Re: [Asterisk-Users] PRI signaling experts please help
Already am using this option. On 8/25/05, Jens von Bülow [EMAIL PROTECTED] wrote: Hi Eric, Don't you need to use out-of-band PRI signaling... From /etc/asterisk/zapata.conf snip ; PRI Out of band indications. ; Enable this to report Busy and Congestion on a PRI using out-of-band ; notification. Inband indication, as used by Asterisk doesn't seem to work ; with all telcos. ; ; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT ; inband: Signal Busy/Congestion using in-band tones priindication = outofband /snip Hope that Helps Jens -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Bishop Sent: 25 August 2005 09:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] PRI signaling experts please help Hi all, Our Asterisk box sends calls outbound via either SIP (through our VoIP provider) or an E1 PRI (directly connected via a TE410P). When we dial a number that is engaged via our VoIP provider we get the following on the Asterisk console (numbers and IP addresses changed to protect the innocent): -- Called [EMAIL PROTECTED] -- Got SIP response 486 Busy here back from 123.123.123.123 -- SIP/sip-outbound-af71 is busy == Everyone is busy/congested at this time This is what we want as it then send the call to priority n+101 and we can handle it any way we want from there. However if the outbound call is made via the PRI (Zap channel) to an enaged number it simply plays an enaged signal to the caller and never progresses to priority n+101. Anyone have any suggestions? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ignorepat not working - what might I have done?
ignorepat only works for analong phones connected to FXS modules.
Steve Maroney
On Fri, 26 Aug 2005, Mason Loring Bliss wrote:
Hey, all. I have the following, and
ignorepat = 9
; Testing - access to telco1/FXO
; XXX
exten = _9.,1,Dial(SIP/outboundfxo/${EXTEN:1},20)
exten = _9.,2,Hangup
Unfortunately, once I hit 9 on a connected phone, I do *not* get a dial
tone back.
Can someone suggest what I might have done wrong?
Thanks!
--
Mason Loring Bliss [EMAIL PROTECTED]http://blisses.org/
I am a brother of jackals, and a companion of ostriches. (Job 30 : 29)
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[Asterisk-Users] Asterisk on VMWare 4.5, Error Ouch ... error while writing audio data
I' m using Asterisk 1.09 on an virtual pc (VMWare 4.5) for testing. I can make calls from a Softphone to softphone, Hardphone to Softphone and so on. I can hear both RTP Streams. But when I call prompst on Asterisk I can hear nothing. RTP Stream goning from Phone to Asterisk but not the other way. I I start the PBX for console I got an error [EMAIL PROTECTED] root]# Ouch ... error while writing audio data: : Broken pipe ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ztmonitor values when zap channel is onhook
We just replaced our old system at work with Asterisk. We use a TDM card with 3 FXO ports. In tuning my gains, I discovered that the on-hook rx levels do in fact waver a little bit. Line 1 is right around 145, Line 2 is around 235, Line 3 is around 380, and Line 4, which doesn't go to a POTS line but instead to a PhoneLabs Cell Socket thingy, is around 100. These numbers sometimes spike upwards by 30% or so, but not too often. So after feeling quite happy about the rx and txgain levels set at my workplace, I did the same steps at home, but in the process noted that my on-hook levels are closer to 3500. They waver a bit, but not more than 30% it seems. This is on a X100P, which I've always suspected was sold to me mis-labeled (I think it's a clone instead of a real X100P). Also, I have a DAML (sometimes also called a concentrator?) installed outside my house, to split the copper lines coming to my house with neighbors. ( I live in a trailer court and personal copper pairs are at a premium I guess. Had to switch away from 56k so long ago 'cause it wasn't 56k anymore...) but anyway the point is that the DAML might be causing the rx noise if it isn't my X10(0|1)P. By way of summary, I feel based on my own experience that the numbers you are seeing while on-hook must be perfectly acceptable and well within normal, usable limits. I think I'll order a real X100P or maybe a tdm with one FXO port to replace my junk and post back to the list with my results. Moj ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: SV: [Asterisk-Users] Cisco and protocol application invalid
Bjorn, Save yourself the trouble with the console cable. Take a spare PC, install [EMAIL PROTECTED], setup the included DHCP server, install the appropriate files in the /tftpboot directory, and plug the PC and the phone into the same switch. Voila! Tom On Aug 26, 2005, at 12:48 PM, Bjørn Ove Kristiansen wrote: Thanks for the suggestions everyone! The thing is, when I run tcpdump, this phone never really requests anything as far as I can see. The IP-address serving is not handled by the Asterisk box, which is on its dedicated IP-address, but by a consumer type SMC Barricade 4-port router. There's no option to define boot tftp on this device. I'm thinking of getting myself a console cable for the phone, and see if I'm able to log in and set the variables using that. Don't know if these cables can be purchased in stores or if they'll have to be hand made.. My biggest problem is that I simply do not know which IP-address the phone is searching for - if its searching for any IP address at all. Thanks again, have a great weekend. Bjorn -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Tom Rymes Sendt: 16. august 2005 16:47 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: [Asterisk-Users] Cisco and protocol application invalid On Aug 15, 2005, at 11:28 AM, Bjørn Ove Kristiansen wrote: Hello! The issue is simply that I don't know which IP address the phone tries to connect to. I am not very familiar with dhcpd (never put it up by hand), so I'm not sure how the below would help me, but from what I can tell, I still need information on which IP-address the phone is trying to find its tftp on, right? Bjorn [snip] Michiel van Baak wrote: I have put this in my dhcpd.conf to make sure my cisco phones connect to my TFTP server: server-name 192.168.2.1; I'd be surprised if that worked... the server name is for.. um.. the name of the server :) Try: option tftp-boot-server code 150 = ip-address; option tftp-boot-server 192.168.44.3; [snip] Bjorn, The phone will use whatever TFTP server your DHCP server tells it to use. That is what the above line option tftp-boot-server is supposed to do, it tells your DHCP server what address to give your phone. If your DHCP server does not give the phone an address, and you have not specified one manually through the phone's settings interface, then it will (in my experience) default to the same address as your DHCP server. What are you using for a DHCP server? Can you enter the settings interface of the phone if you leave the ethernet disconnected? (If so, manually specify the TFTP server address.) Tom___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.10/73 - Release Date: 15.08.2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.15/82 - Release Date: 25.08.2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
