[Asterisk-Users] Anyone with VIP-450
Hi all, While I am making my VoIP gateway to fix G.711, I just want to know who has done Asterisk with Planet VIP-450T DTMF out-of-band relay before... The supporting document says "DTMF relay uses SIP specification", it sounds like it support SIP-INFO, but I'm not sure. Any help will be pleased~ Best Regards, Jason Chan, Hong Kong No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.13.13/197 - Release Date: 9/12/2005 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Wildcard TDM2400P: comments
We are also looking for analog port for fax and dialup modem. Yusuf, would you pls descript what stability issues you had with TDM400P? We are thinking about using TDM400P or Voicetronix OpenPCI-8S. Cheers, Isaac Well, we used the TDM400P a while back, obviously things might have been fixed sinc then. The issues I had were: the card just hanging the machine, or the card just stop taking calls. Can you define a LOT of pots line? Have you considered a channel bank. Here I'm running an ADTRAN 750. It's painless. You just need 1 T1 interface card for 24 lines. Jacques round about 30 pots lines. channel bank might be an option thanks Jacques yusuf wrote: Hi all, we have the need for alot of plain analog lines. We thinking of buying the new Wildcard TDM2400P. Does anybody have any comments with using this card, with any version of Asterisk, (maybe ill make this one Asterisk 1.2.x). I have had some stabilty issues using the 4 TDM400P. What about this new TDM2400P??? thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Starting RTP with Dial and MusicOnHold
Hello, Do you try Answer() and then Dial(SIP/xyz,,m)??? Exten = ???,1,Answer() Exten = ???,2,Dial(SIP/xyz,,m) You need to answer the call before you can hear music on hold. Pedro Nunes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Clauson Sent: quinta-feira, 15 de Dezembro de 2005 4:45 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Starting RTP with Dial and MusicOnHold Hi, I'm trying to get Asterisk working with a supplier's Cerpack switch and everything is working except audio ringback for calls coming from Cerpack to Asterisk. The Cerpack switch only does out of band progress indication (seems a bit strange for SIP to SIP calls?!) so I've spent the last two days trying to find a way to force Asterisk to send an RTP stream to Cerpack for ring back. Theoretically the Dial command with the m option looks to be exactly what I need: Dial(SIP/xyz,,m) This should play musiconhold back to the caller and in my case I just took a recording of the PSTN tones I wanted to play and created a musiconhold class for them. The command will work correctly when dialled from a SIP phone connected to Asterisk but not for calls coming from Cerpack. As far as I can tell this is because Asterisk won't initiate the RTP stream and waits for a packet from the client before starting to play the musiconhold, perhaps assuming the connection is not available until it gets a packet. In this case Cerpack isn't sending a packet so no audio is heard until the call is answered. Has anybody seen anything like this before? Thanks, Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AoC (Advice of Charge)
Does Asterisk support Advice of Charge? I was told that my Telco sends me billing signalization that way, and I wonder can I use it? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fax detected, but no fax extension
OK,
Is Asterisk able to switch incoming calls according to
fax or voice to the right extension .
Which function detect incoming signal ?
Regards
H.G
--- Colin Anderson [EMAIL PROTECTED]
a écrit :
You need an extension called fax in your [fax]
context like this:
[fax]
exten = fax,1,Goto(macro-faxreceive,s,1)
hth
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 14, 2005 3:18 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Fax detected, but no fax
extension
Hello,
I get this message when i send fax Fax detected,
but
no fax extension.
I read mailing list .
Can we solve this ?
my conf :
=PSTN==(fxo)-Asterisk-(fxs)==Hylafax==SMTP/POP3
server
zapata.conf:
context=fax
faxdetect=both
signalling=fxo_ks
group=2
channel = 2
extension.conf
[fax]
exten = 80,1,Dial(Zap/2)
ingnorepat = 0
include = outgoing-pstn
[outgoing-pstn]
exten = _0,1,Dial(Zap/g1/${EXTEN:1})
exten = _0.,1,Dial(Zap/g1/${EXTEN:1})
Regards
H.G
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[Asterisk-Users] QueueMetrics 1.0 rc 1 out today
Hello list, I am pleased to tell you that we have released a new version of QueueMetrics. This time QueueMetrics comes of age, as we have almost reached version 1.0. :-) This version is the result of a feature-freeze of version 0.9.7, and vastly improves memory efficiency, letting you analyze as much as two to four times the number of calls using the same amount of RAM. QueueMetrics is strongly focused to serve the largest commecial call-centers based on Asterisk, and is very often being installed on 100+ agents systems. Pause handling was improved in order to support even the weirdest cases of agents pausing in/out when not logged on. We are confident that you'll love this version! A complete list of improvements can be found here: http://queuemetrics.loway.it/news.jsp The latest version of QM can be downloaded from http://queuemetrics.loway.it/download.jsp QueueMerics is free to use and experiment for smaller call centers, home users and general Asterisk hackers. Larger installations can ask us for a free trial key from http://queuemetrics.loway.it/sendDemoLicence.jsp -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Starting RTP with Dial and MusicOnHold
-Original Message- From: Pedro Nunes [mailto:[EMAIL PROTECTED] Sent: 15 December 2005 08:59 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Starting RTP with Dial and MusicOnHold Hello, Do you try Answer() and then Dial(SIP/xyz,,m)??? Exten = ???,1,Answer() Exten = ???,2,Dial(SIP/xyz,,m) You need to answer the call before you can hear music on hold. Hi Pedro, What you suggest would work but is no good as anybody calling our numbers would be charged for the call. The Dial(,,m) command can play MusicOnHold without answering the call, I know I've tested it ;-). In this case I just need to give the RTP a kick start or something, the console reports the MusicOnHold has started playing but there is no RTP. Thanks, Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] background music...
Hi, list! I have a strange problem. I already tried to get help from the lsit, but I hadn't luck enough... This is the scenario: a GSM mobile phone calls to asterisk. After answering, if asterisk plays background music, when the caller speaks, the music becomes unhearable, too noise, it's not possible to distinguish it from noise. I don't know how to handle this. I've tried several codecs por playing the music, but i don't manage... I want the music to be hearable even if the caller speaks... Did anyone have a similar problem? Can anyone help me? thanks -esteban- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to tell if Authenticate failed without using j in 1.2
How can one tell if Authenticate fails in 1.2 without using the j option? There appears not to be a STATUS variable set by this app. -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with mgcp
Please, somebody can help with mgcp problems? My emterprise is ready to remove the asterisk solution and replace it with an old hardware pbx due to the lot of problems that has asterisk. I'm assuring them the problems is with mgcp but they beleave the problems are of asterisk. (using asteriskatmome 2) -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Firewall Ports forward
hi all. i have my asterisk with a 192.168.0.1 address which ports i need to forward in my firewall to connect remote xten clients and make calls? thsnk -- .- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] again - show queue info
Hello list, I am comparing results of "show queue myqueue" with data from /var/log/asterisk/queue_log and have some doubts about it. When I run "show queue myqueue", I get a value of 3 for the number of abandoned. When I check the queue_log file, I have 3 calls with status "EXITWITHTIMEOUT". This way I have realized that A means "unAnswered" and not actually "Abandoned". ( I GUESS EXITWITHKEY calls would also increment the value of A). My queue has a relatively short time out (45 secs) and then the caller redirected to a voicemail. I have developed a real time monitoring application that is not handling events yet, it is simply sending manager commands and printing out the results, and my call center managers are not satisfied with the information I am currently displaying, so, handling event is certainly the best way to accomplish my goals. Does any body else have comments about this statistics, and ways of showing good real time information for call center managers? Thank you Dov -- exten = cobrancainfo,1,Answerexten = cobrancainfo,2,Queue(infocadastrais|tT|||45)exten = cobrancainfo,3,Wait(3)exten = cobrancainfo,4,VoiceMail(u501)exten = cobrancainfo,5,Hangup -- lv09*CLI show queue infocadastraisinfocadastra has 0 calls (max unlimited) in 'leastrecent' strategy (0s holdtime), W:0, C:0, A:3, SL:0.0% within 45s Members: Agent/5132 (Unavailable) has taken no calls yet Agent/4952 (Not in use) has taken no calls yet Agent/2732 (Unavailable) has taken no calls yet Agent/2462 (Unavailable) has taken no calls yet No Callers 1134644282|1134644268.462750|infocadastrais|NONE|ENTERQUEUE||pabx1134644328|1134644268.462750|infocadastrais|NONE|EXITWITHTIMEOUT|11134644358|1134644344.463504|infocadastrais|NONE|ENTERQUEUE||pabx1134644410|1134644344.463504|infocadastrais|NONE|EXITWITHTIMEOUT|11134644470|1134644456.464234|infocadastrais|NONE|ENTERQUEUE||pabx1134644516|1134644456.464234|infocadastrais|NONE|EXITWITHTIMEOUT|1 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 PBX linked via internet
www.voip-info.org is one stop ref. for asterisk help. yes its more then possible. You can put asterisk server at your place and make your friend login to your ast box. you can make your friend a sip account use xlite ( windows based softphone ) at your friends place. IMHO you need to buy fxo card to plug your telco or vonage line into ast box. yes thenyour friend can use your vonage or telco line to dialout, and you can make dialplan which actually let your friend chose how to dial and from which line. regards, Umair bari On 12/15/05, Ugo Bellavance [EMAIL PROTECTED] wrote: Hi,I think it is the first time I post on this list,I went throughthe few couple of hundred last messages on the list and couldn't find an answer.I also read the FAQ.I'd need some advice... here is what I plan to do.I've neverinstalled asterisk so I know I need to do some testing.I don't mindbuying equipment, but I don't want to overspend, especially in the test phase.I planned on using [EMAIL PROTECTED] on old boxes that I can get forless than 100$ (P3 500/128 MB) or astlinux on pcengines WRAP machines(or even astlinux on the P3).I just bought a WRAP that is going to eventually replace my netgear home router, but I could use the WRAP fortesting.There is no hurry to replace the netgear.What I want to do is have one asterisk PBX here and one at myfriend's house.We are both with the same ISP, cable modem, with good bandwidth.My friend lives ~ 25 Km from me.- The incoming phone linewould be at my house, but I'd like to have an extension going to myfriend's.I am testing traffic shaping rules with my firewall right now, so this is likely to be in the plan as well.At home I have 2 lines: one regular telco line and one VoIP linefrom Vonage (with a motorola phone adapter).The vonage has unlimitedlong distance USA/Canada (I'm in canada btw).Here are my questions: - Is that possible?Any links to howtos?- Could I connect both of my lines on the pbx?- Can my friend dial through my vonage line to make long-distance calls?Can he actually choose?- I'd like to avoid buying digium cards at least for the testing phase... I know I'll probably need one to use my telco line or the IPline behind the phone adapter, but if I can avoid buying one for myfriend's PBX, it would be great. Can I use (free or inexpensive)softphones with asterisk? Thanks in advance,--Ugo- Please don't send a copy of your reply by e-mail.I read the list.- Please avoid top-posting, long signatures and HTML, and cut theirrelevant parts in your replies. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I don't want ilbc, i just want G.711
in your sip.cong [general] contexts put disallow=all allow=ulaw allow=alaw and in your sip user, use disallow only ONCE, that is disallow=all allow=ulaw allow=alawhope this helps. regards, Umair bari On 12/15/05, Jason Chan (jasonOfficial) [EMAIL PROTECTED] wrote: Hi there,I am writing to ask about how to fix the codec to G.711 ONLY.Actually what I am doing is, try to use DTMF when the POTS phone call hasdirected to Asterisk via Planet VIP-450 FXO Port, but this gateway just simply doesn't support RFC2833 nor SIP-INFO. The only method I can use isInband DTMF. I know it only support G.711, but I DID disallow others andmake it work only with G.711. But the problem is, although I disallow all other codecs, ilbc still itching me...[extensions.conf][852]username=HKGWserect=blahtype=friendhost=dynamicnat =yescanreinvite=nodisallow=alldisallow=ilbcallow=ulawdtmfmode=inband (P.S. I don't use REINVITE simply because I need the asterisk to be amedia gateway cause the gateway is inside NAT behind the Asterisk)Whenever I try to pass DTMF from phone to Asterisk via that gateway, I got such messages:Dec 14 23:35:32 WARNING[10958]: dsp.c:1422 ast_dsp_process: Inband DTMF isnot supported on codec ilbc. Use RFC2833Dec 14 23:35:32 WARNING[10958]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?How come!? I DID DISALLOW them, but it keeps bugging me= 192.168.2.3 852 79f9e0-c0a8 00101/1 ulaw No Rx:ACK1 active SIP channel*CLI sip show channel 79 * SIP Call Direction: Incoming Call-ID: [EMAIL PROTECTED] Our Codec Capability: 4 Non-Codec Capability: 0 Their Codec Capability: 261 Joint Codec Capability: 4 Format ulaw Theoretical Address: 192.168.2.3:5060 Received Address: 192.168.2.3:5060 NAT Support: Always Audio IP: 192.168.2.1 (local) Our Tag: as737358ce Their Tag: 3a53f3e1-bbfcafe6d5c SIP User agent: Username: 852 Peername: 852 Original uri: sip:[EMAIL PROTECTED]:5060 Caller-ID: elite Need Destroy: 0 Last Message: Rx: ACK Promiscuous Redir: No Route: sip:[EMAIL PROTECTED]:5060 DTMF Mode: inband SIP Options: (none)==Previously I installed 1.0.3 in same machine, but I overwrite all files with 1.2.1.. does it cause a trouble?Can anyone figure out what is the problem? ==Thanks very much for your help!Best regards, Jason Chan, Hong KongNo virus found in this outgoing message.Checked by AVG Free Edition.Version: 7.1.371 / Virus Database: 267.13.13/197 - Release Date: 9/12/2005___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Fwd: Re: [Asterisk-Users] Re: [helpp] Problem in astersik]
Dear Talat,
if you are trying to connect from within your LAN,
put nat=no and then try againregards,
Umair bari
On 12/14/05, Talat Ishtiaq [EMAIL PROTECTED] wrote:
-- Forwarded message --From: Talat Ishtiaq
[EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comDate: Wed, 14 Dec 2005 15:38:08 +0500
Subject: Re: [Asterisk-Users] Re: [helpp] Problem in astersikHi GuysAfter your guies replies now i have changed the machine .But this time iget little different problemi made following chnages in
sip.conf[901]context=fromsiptype=friendusername=901secret=901callerid=Test2 901host=dynamicnat=yescanreinvite=nodisallow=allallow=ulawdtmfmode=rfc2833
callgroup=3pickupgroup=3qualify=1000;[902];context=fromsip;type=friend;username=902;secret=902;callerid=Test3 902;host=dynamic;nat=yes;canreinvite=no
;disallow=all;allow=ulaw;dtmfmode=info;callgroup=3;pickupgroup=3;qualify=1000in extension.conf[fromsip]exten = s,1,Answer( )exten = _9XX,1,Dial(SIP/${EXTEN},100,tr)
exten = _5XX,1,Dial(SIP/${EXTEN},100,tr)exten = h,1,Hangupexten = t,1,Hangupexten = i,1,HangupNowAsterisk 1.0.9, Copyright (C) 1999-2004 Digium.Written by Mark Spencer
[EMAIL PROTECTED]=[ BootingDec 14 15:23:05 WARNING[3478]: chan_oss.c:257sound_thread: Read error on sound device: Resource temporarily
unavailable.Dec 14 15:23:07 WARNING[3478]: chan_skinny.c:2587 reload_config: Unable to get our IP address, Skinny disabled ]
Asterisk Ready.*CLINow from xpro lite software after configuring it for my machine when itry to connect to my machine i am unable to get connection it saysunable to connect contact your network
administratot.Althoug i am thenetwork adminPlz tell me what to doRegardTalatOn Mon, 2005-12-12 at 06:40 -0500, Steven wrote: /var/log/asterisk/full text file may give you a more specific error.
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[Asterisk-Users] How to change the Dial command H option to ## ?
I want to use '##' to terminate a call instead of the '*' used by the Dial command's H option. Is there a way to change the key or use another option to achieve the same effect? /Obelix This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firewall Ports forward
506045691-2 On 12/15/05, Pablo Allietti [EMAIL PROTECTED] wrote: hi all. i have my asterisk with a 192.168.0.1 addresswhich ports i need to forward in my firewall to connect remote xten clients and make calls?thsnk--.-___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 17, Issue 89
I did the following s,1,Background(blablabla) s,2,Read(VARIABLE||1) ; accepting only one digit (1 to accept call, anything else to hangup) That's not the right approach. Do something like his: [confirmcall] exten = s,1,Background(blablabla) exten = 1,1,Goto(accept_call_context,s,1) exten = t,1,Hangup exten = i,1,Hangup Thanks Luki. I was just following the example in the Wiki, on the Dial() cmd page. But now that I think of it, it does make more sense to use your approach. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-3000: Dual Registrations?
I found the problem, finally. Or perhaps better put, I found a way to make it work. I'm sending to list in case someone else finds him/herself in the same spot someday. It turns out that the sip.conf context name and the registration username must be the same. Once I had done that for both the user and peer stanzas in sip.conf, things worked just fine. Argggh. If I'm reading my files properly, that is a requirement for SIP but not for IAX? And I wonder if it's something in the protocol or something about the Asterisk SIP configuration method. I believe its been that way since I started with * a couple of years ago. There are many other sip items in similar shape that one might consider irregularities as well. We're all looking forward to improvements as * moves forward. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to change the Dial command H option to ## ?
its build into the codes, IMHOfor replacing * with ## you need to hack asterisk source code, I cant think of anyother way. regards, Umair bari On 12/15/05, Obelix [EMAIL PROTECTED] wrote: I want to use '##' to terminate a call instead of the '*' used by the Dialcommand's H option. Is there a way to change the key or use another option to achieve the sameeffect?/ObelixThis message was sent using IMP, the Internet Messaging Program. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: how to forward call within office
hi all, I am newbie to asterisk. I have installed asterisk sever on debian. I have also installed 4 X-Lite phones on four PCs. My all phones and asterisk server is working properly. Now i want to implement call forwariding facility on my server. I want to forward the call made by phone-1 to phone-2 , to another phone i.e. phone-3. Within office premises is it possible to forward calls? if yes, How can i implement this? what configurations files i have to change? I have one othere query. i want to knw, can i use "s" extension for my office pbx? I mean i dont have any PSTN connection to my asterisk server. I m using asterisk just as a internal gateway. so how can i use "s" extension? If i m not wrong "s" extension is used when call comes from PSTN line. Am i right? or m i making a wrong assumption ? Thanks tejas Yahoo! Shopping Find Great Deals on Holiday Gifts at Yahoo! Shopping ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 : which versions recommended for asterisk 1.2?
Hi, Which oh323 versions recommended for asterisk 1.2? Where can I get them? I use: $ ls -la total 7172 drwxr-xr-x 8 john john4096 Dec 15 15:12 . drwxr-xr-x 17 john john4096 Dec 15 13:27 .. drwxr-xr-x 5 john john4096 Sep 20 15:53 asterisk-oh323-0.7.3 -rw-r--r-- 1 john john 93440 Sep 20 15:55 asterisk-oh323-0.7.3.tar.gz drwxr-xr-x 8 john john4096 Dec 15 13:41 openh323 -rw-r--r-- 1 john john 2555677 Sep 22 19:52 openh323-Janus_patch4-src-tar.gz -rw-r--r-- 1 john john 2343354 Sep 22 19:50 openh323_1.12.2.tar.gz drwxr-xr-x 8 john john4096 Dec 15 14:33 pwlib -rw-r--r-- 1 john john 229 Sep 22 19:54 pwlib-Janus_patch4-src-tar.gz -rw-r--r-- 1 john john 1085203 Sep 22 19:53 pwlib_1.5.2.tar.gz On asterisk-oh323-0.7.3 compiling I got error: $ make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/home/john/devel/oh323/asterisk-oh323-0.7.3/wrapper' ./check_ver /home/john/devel/oh323/pwlib pwlib ./check_ver /home/john/devel/oh323/openh323 openh323 g++ -Wall -felide-constructors -x c++ -Os -DP_USE_PRAGMA -ffunction-sections -fdata-sections -D_REENTRANT -Wall -fPIC -I/home/john/devel/oh323/pwlib/include -DPTRACING -I/home/john/devel/oh323/openh323/include -DHAS_OSS -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\ -I/home/john/devel/oh323/pwlib/include -I/home/john/devel/oh323/openh323/include -I/home/john/devel/oh323/openh323/include/openh323 -I../asterisk-driver -c wrapendpoint.cxx -o wrapendpoint.o wrapendpoint.cxx: In member function `virtual BOOL WrapH323EndPoint::OpenAudioChannel(H323Connection, int, unsigned int, H323AudioCodec)': wrapendpoint.cxx:800: error: syntax error before `)' token wrapendpoint.cxx:800: error: `PIsDescendant' undeclared (first use this function) wrapendpoint.cxx:800: error: (Each undeclared identifier is reported only once for each function it appears in.) wrapendpoint.cxx:801: error: syntax error before `)' token What's wrong? -- Thanks, Eugene Prokopiev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Small explanation of txgain rxgain statement please
Hi, I was just looking at: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html regarding echo canceller tuning, and I noticed the statement Most people find that they need an rxgain level around 8.0 to have good echo cancellation. The txgain setting varies from installation to installation. Which feels a bit wrong :) Could someone explain why increasing the gain on the inbound zap leg (rxgain) would improve echo cancellation? Of have I misunderstood the roles and meanings of rxgain and txgain? Many thanks Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] EXITWITHQUEUE on queue_log
Is there way to disable the possibility of a user leave a queue by pressing a queue? I have several occurences of EXITWITHKEY in my queue_log that shouldn't occur... 1134642743|1134642524.462637|cobranca|Agent/5230|TRANSFER|350|default1134642743|1134642524.462637|cobranca|NONE|EXITWITHKEY||1 1134646015|1134645980.464421|cobranca|NONE|ENTERQUEUE||343 1134646035|1134645980.464421|cobranca|Agent/5100|CONNECT|20 1134646171|1134645980.464421|cobranca|Agent/5100|COMPLETEAGENT|20|1361134646171|1134645980.464421|cobranca|NONE|EXITWITHKEY||1 Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] screen safe_asterisk does'nt spawn asterisk
screen -d -m asterisk -vvvcng works well for me, but I'd prefer to run safe_asterisk in production anyway 'screen -d -m safe_asterisk' spawns no asterisk processes, anyone knows the reason ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sharing a line w/multiple extensions
I'd like to configure Asterisk so an incoming call from one POTS line is shared amongst multiple extensions - both SIP and analog. i.e. If one SIP phone answers the call, another SIP or analog extension phone can pick up and join the conversation. How do I configure this? Is it all in extensions.conf? Asterisk is not a key system. It does not behave this way. What do you mean by 'another SIP phone can pick up (...) the conversation'? Exactly what would the SIP phone user do to accomplish that? Think residential installation where someone picks up the phone in one room but someone in another room wants to join the conversation. Ideally, I'd like to have Line 1 on every phone (SIP or analog) behave this way. Another poster pointed out a good potential approach using meetme. When an incoming call comes in, it dials all SIP + analog phones. When someone picks up (don't know how I can detect this), it could transfer both parties to a meetme room. When additional extensions pickup, they go to the meetme room. When everyone hangs up, the call ends. Can this be done? There might be a way for you to address your objective depending upon exactly what you're trying to do. The previous responses to your question _assume_ that each room in your case has a pbx extension (regardless of whether its a sip or analog phone). If their assumption is correct, then the responses are correct. However, if you want to use your existing analog phones and you group them together, several analog phones can share a single extension and those phones in the group can pick up and join the conversation whenever they want. Think in terms of using something like a Sipura sip adapter (or the equivalent from other vendors), and connecting all analog phones within your defined group to the rj11 analog jack of the adapter. For example, I have four analog pstn lines and multiple iax connections to various itsp's and clients. One of the analog pstn lines is a house line and connects directly to * via a TDM04b card. When an incoming call occurs on that line, it rings multiple sip phone/adapters. One of those happens to be a Sipura spa3000 that has most of the analog house phones attached. Anyone one of those phones can answer the call, others can join in, etc. The approach can work if you can define specific groups of interest such as kids vs adults, sales vs support, home vs business, etc. Combine that approach with carefull selection of analog phones (those with some form of line in use LED), and you end up with an approach that sort of looks like a poor-man's key system behind a pbx. Pay attention to the features within the sip adapter (eg, Sipura) and you're likely to find additional options that might address your needs. All depends upon exactly what it is that you're trying to engineer. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: how to forward call within office
hi all, I am newbie to asterisk. I have installed asterisk sever on debian. I have also installed 4 X-Lite phones on four PCs. My all phones and asterisk server is working properly. Great start. Now i want to implement call forwariding facility on my server. I want to forward the call made by phone-1 to phone-2 , to another phone i.e. phone-3. Within office premises is it possible to forward calls? if yes, How can i implement this? what configurations files i have to change? You can do this on phones if it is supported and all of your call flow logic is going to be in extensions.conf or some file included. I have one othere query. i want to knw, can i use s extension for my office pbx? I mean i dont have any PSTN connection to my asterisk server. I m using asterisk just as a internal gateway. so how can i use s extension? If i m not wrong s extension is used when call comes from PSTN line. Am i right? or m i making a wrong assumption ? http://www.voip-info.org/wiki-Asterisk+s+extension Note that most of the type of calls that I deal with have a known number called, so S extension would never apply. That is, with a PRI, IAX DID provider... Thanks tejas Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to change the Dial command H option to ## ?
Obelix wrote: I want to use '##' to terminate a call instead of the '*' used by the Dial command's H option. Is there a way to change the key or use another option to achieve the same effect? Application map in features.conf assigning ## to Hangup() ? Maybe :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan-capi avm b1 and capi.conf problems
Hello everybody. First, pardon my bad english. I have installed last asterisk and chan-capi-cm-0.6.1 debian distro and kernel 2.6.13.4 Hardware is well detected, output of lspci: :00:0c.0 Network controller: AVM Audiovisuelles MKTG Computer System GmbH B1 ISDN I downloaded b1.t4 file and i follow several guides from voip-info.org and searched with google, i also have /etc/isdn/capi.conf file: # card file proto io irq mem cardnr options #b1isa b1.t4 DSS1 0x150 7 - - P2P b1pci b1.t4 DSS1 - - - - #c4 ... The isdn card is connected point to multipoint and i can do capiinit start, stop and so on: Dec 15 11:50:56 pbx kernel: capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs) Dec 15 11:50:56 pbx kernel: b1dma: revision 1.1.2.3 Dec 15 11:50:57 pbx kernel: b1pci: PCI BIOS reports AVM-B1 at i/o 0x2000, irq 18 Dec 15 11:50:57 pbx kernel: kcapi: Controller 1: b1pci-2000 attached Dec 15 11:50:57 pbx kernel: b1pci: AVM B1 PCI at i/o 0x2000, irq 18, revision 2 Dec 15 11:50:57 pbx kernel: b1pci: revision 1.1.2.2 Dec 15 11:50:58 pbx kernel: b1pci-2000: card 1 B1 ready. Dec 15 11:50:58 pbx kernel: b1pci-2000: card 1 Protocol: DSS1 Dec 15 11:50:58 pbx kernel: b1pci-2000: card 1 Linetype: point to multipoint Dec 15 11:50:58 pbx kernel: b1pci-2000: B1-card (3.11-03) now active Dec 15 11:50:58 pbx kernel: kcapi: card 1 b1pci-2000 ready. The main problem is that can load asterisk (asterisk -vvvcg) but i get just one error message and, off course isdn do not work: Dec 15 11:13:25 ERROR[4887]: chan_capi.c:4835 load_module: Unable to load config capi.conf, CAPI disabled I have capi.conf file as i show you before... Can someone help me please? Pardon if it is a nonsense, i am not an expert Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hint on Zap channels
Hi all has anyone an working example of a hint-entry with a Zap-Channel ? I've got hint working with SIP and SCCP but Zap doesn't seem to work ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hint on Zap channels
[EMAIL PROTECTED] ha scritto: has anyone an working example of a hint-entry with a Zap-Channel ? I've got hint working with SIP and SCCP but Zap doesn't seem to work Fixed in current CVS 1.2 and HEAD older versions have a case sensitivity issue so you have to write it in the right way this one works exten = 1, hint, Zap/1 this one does not work exten = 1, hint, ZAP/1 this one does not work exten = 1, hint, zap/1 Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail cutting out
I have a user that when they go to leave a voicemail message for another user it cuts them off because the talk to quiet. When they talk loud it works fine. Is there a way to fix this ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sharing a line w/multiple extensions
On Thu, Dec 15, 2005 at 06:11:07AM -0600, Rich Adamson said: I'd like to configure Asterisk so an incoming call from one POTS line is shared amongst multiple extensions - both SIP and analog. i.e. If one SIP phone answers the call, another SIP or analog extension phone can pick up and join the conversation. How do I configure this? Is it all in extensions.conf? Asterisk is not a key system. It does not behave this way. What do you mean by 'another SIP phone can pick up (...) the conversation'? Exactly what would the SIP phone user do to accomplish that? Think residential installation where someone picks up the phone in one room but someone in another room wants to join the conversation. Ideally, I'd like to have Line 1 on every phone (SIP or analog) behave this way. Another poster pointed out a good potential approach using meetme. When an incoming call comes in, it dials all SIP + analog phones. When someone picks up (don't know how I can detect this), it could transfer both parties to a meetme room. When additional extensions pickup, they go to the meetme room. When everyone hangs up, the call ends. Can this be done? There might be a way for you to address your objective depending upon exactly what you're trying to do. The previous responses to your question _assume_ that each room in your case has a pbx extension (regardless of whether its a sip or analog phone). If their assumption is correct, then the responses are correct. However, if you want to use your existing analog phones and you group them together, several analog phones can share a single extension and those phones in the group can pick up and join the conversation whenever they want. Think in terms of using something like a Sipura sip adapter (or the equivalent from other vendors), and connecting all analog phones within your defined group to the rj11 analog jack of the adapter. One system I found that works well in a home environment is using a two-line, multi-handset cordless phone system. Run 2 analog ports to the base station, and this handles most home needs. Two users can make or receive calls, join existing calls, etc rather easily. The dial plan is set so that either line makes outgoing calls over a VoIP service, line 2, or whatever, so that the main incoming line is always available to receive calls. The home office has a Polycom 601 with it's own lines and dial plan logic, plus the fact that the polycom user is much more likely to know how to answer, transfer, park, etc. Wife proofing a * system is non-trivial and takes careful planning. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: how to forward call within office
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I have one othere query. i want to knw, can i use s extension for my office pbx? I mean i dont have any PSTN connection to my asterisk server. I m using asterisk just as a internal gateway. so how can i use s extension? If i m not wrong s extension is used when call comes from PSTN line. Am i right? or m i making a wrong assumption ? You can use s extension in your local SIP phonecalls. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sharing a line w/multiple extensions
phones. When someone picks up (don't know how I can detect this), it could transfer both parties to a meetme room. When additional extensions pickup, they go to the meetme room. When everyone hangs up, the call ends. Can this be done? Probably, but it would take some very creative dialplan programming and an external application to transfer the parties into a meetme room. You will not get 'pickup' behavior on the SIP phones regardless, they will have to press a speed dial button which would attempt to join the meetme. In other words: you can get there, but it will _not_ behave like a key system, and people will expect it to, so they will be frustrated when it doesn't. We've been down this road many times before, and many Asterisk installations have been taken out because the installers thought they could achieve key system behavior (or retrain the users) but failed. If you want to try, feel free... I'm only telling you what has happened before :-) Thanks. That's very helpful because being new to Asterisk, I don't know the history of what people have attempted to use Asterisk for. It's unfortunate that there's no way to do it because it sounds like others are looking for this same functionality. I wonder what it would take to implement this in Asterisk natively. Does Digium take feature requests? Certainly, this would have appeal for residential systems. Just a couple of comments on the subject of key systems verses pbx's The traditional pbx (from years ago) implemented exactly what Kevin mentioned above. Just about every company that deployed a pbx back then also had several key systems attached to their pbx. The key systems were typically limited to executives and their assistants (secretaries back then) primarily due to the additional cost of the older key systems. The traditional pbx vendors (back then) would always use the same words that Kevin used, emphasizing the differences between key systems and pbx's. However, many of the pbx manufacturers finally realized they were loosing revenue due to those limitations, and began implementing key-system-type functions in their pbx's. They were not trying to address the key system market, but rather make their pbx products more valuable from a user's perspective. Those that are influencing or controlling the direction of asterisk haven't learned that lesson as yet, partially because of the lack of functionality in the sip phones themselves and partially because asterisk is being developed through the open source community (limited development resources and no published long term plan). Those individuals that have worked towards developing the sip rfc standards have recognized some of the key system vs pbx needs, and have added to the sip standards. However, it takes a while for the sip phone manufacturers (and voip pbx manufacturers) to implement those standards, and in some cases, the manufacturers purposefully leave out certain functions in their sip products to protect their investments in proprietary products. It certainly is not difficult to visualize how voip switching products (such as asterisk or any of the commercial products) could be oriented towards being a switch and address the needs of key systems, pbx's, and central office switching in the same basic product. All of the same functions are required in each case. Asterisk will get there, it will just take a little longer since there isn't any published long term plan to influence the short term development. (No offense intended to any asterisk individual or group; just the nature of most open source development.) I can also assure you that several large companies (most of those company names likely wouldn't be recognized by many of the readers here) are watching the asterisk development closely, and likely are in fear of various open source products negatively impacting their core business. They will adjust their product development (and plans) in an effort to remain one (or more) steps ahead from a marketing sales perspective. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I don't want ilbc, i just want G.711
And if, for some very strange reason, it doesn't work, use noload at modules.conf ;) Regards, 2005/12/15, Umair Bari [EMAIL PROTECTED]: in your sip.cong [general] contexts put disallow=all allow=ulaw allow=alaw and in your sip user, use disallow only ONCE, that is disallow=all allow=ulaw allow=alawhope this helps. regards, Umair bari On 12/15/05, Jason Chan (jasonOfficial) [EMAIL PROTECTED] wrote: Hi there,I am writing to ask about how to fix the codec to G.711 ONLY.Actually what I am doing is, try to use DTMF when the POTS phone call hasdirected to Asterisk via Planet VIP-450 FXO Port, but this gateway just simply doesn't support RFC2833 nor SIP-INFO. The only method I can use isInband DTMF. I know it only support G.711, but I DID disallow others andmake it work only with G.711. But the problem is, although I disallow all other codecs, ilbc still itching me...[extensions.conf][852]username=HKGWserect=blahtype=friendhost=dynamicnat =yescanreinvite=nodisallow=alldisallow=ilbcallow=ulawdtmfmode=inband (P.S. I don't use REINVITE simply because I need the asterisk to be amedia gateway cause the gateway is inside NAT behind the Asterisk)Whenever I try to pass DTMF from phone to Asterisk via that gateway, I got such messages:Dec 14 23:35:32 WARNING[10958]: dsp.c:1422 ast_dsp_process: Inband DTMF isnot supported on codec ilbc. Use RFC2833Dec 14 23:35:32 WARNING[10958]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?How come!? I DID DISALLOW them, but it keeps bugging me= 192.168.2.3 852 79f9e0-c0a8 00101/1 ulaw No Rx:ACK1 active SIP channel*CLI sip show channel 79 * SIP Call Direction: Incoming Call-ID: [EMAIL PROTECTED] Our Codec Capability: 4 Non-Codec Capability: 0 Their Codec Capability: 261 Joint Codec Capability: 4 Format ulaw Theoretical Address: 192.168.2.3:5060 Received Address: 192.168.2.3:5060 NAT Support: Always Audio IP: 192.168.2.1 (local) Our Tag: as737358ce Their Tag: 3a53f3e1-bbfcafe6d5c SIP User agent: Username: 852 Peername: 852 Original uri: sip:[EMAIL PROTECTED]:5060 Caller-ID: elite Need Destroy: 0 Last Message: Rx: ACK Promiscuous Redir: No Route: sip:[EMAIL PROTECTED]:5060 DTMF Mode: inband SIP Options: (none)==Previously I installed 1.0.3 in same machine, but I overwrite all files with 1.2.1.. does it cause a trouble?Can anyone figure out what is the problem? ==Thanks very much for your help!Best regards, Jason Chan, Hong KongNo virus found in this outgoing message.Checked by AVG Free Edition.Version: 7.1.371 / Virus Database: 267.13.13/197 - Release Date: 9/12/2005___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Starting RTP with Dial and MusicOnHold
Why not to use r option in Dial(SIP/xyz,,r) to simulate the ring? Regards, 2005/12/15, Aaron Clauson [EMAIL PROTECTED]: -Original Message- From: Pedro Nunes [mailto:[EMAIL PROTECTED] ] Sent: 15 December 2005 08:59 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Starting RTP with Dial and MusicOnHold Hello, Do you try Answer() and then Dial(SIP/xyz,,m)??? Exten = ???,1,Answer() Exten = ???,2,Dial(SIP/xyz,,m) You need to answer the call before you can hear music on hold. Hi Pedro,What you suggest would work but is no good as anybody calling our numberswould be charged for the call.The Dial(,,m) command can play MusicOnHold without answering the call, I know I've tested it ;-). In this case I just need to give the RTP a kickstart or something, the console reports the MusicOnHold has started playingbut there is no RTP.Thanks,Aaron___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip configuration for make and receive calls
Hi, I'm having some problems configuring a SIP account from a VOIP service provider named voztele.com. I can configure it to make calls or to receive calls, but not both :) Here is my sip.conf, with it, I can make calls, but not receive. If I comment all of the user voztele, then I can receive call, but can't make... [general] context=default bindport=5060 bindaddr=0.0.0.0 set canreinvite=no register = username:[EMAIL PROTECTED]/100 externip = my_public_ip nat=yes disallow=all allow=alaw allow=ulaw localnet=192.168.1.0/255.255.255.0 [voztele] type=friend username=username fromuser=username fromdomain=serviceprovider.com host=serviceprovider.com secret=secret nat=yes qualify=4000 canreinvite=no reinvite=no dtmfmode=rfc2833 context=default insecure=very ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ChanIsAvail()
Hello,
I configure a asterisk server with tdm400p . I wish
to set chanisavail() in order to allow users or the
hylafax server to dial numbers to pstn . however I
can't write the rules to forward requests to the dial
pattern when channel is available.
I try this however priority 2 fail.
how can i forward requests to outgoing-pstn context ?
exten = s,1,ChanIsAvail(Zap/g1)
exten = s,2,Goto(outgoing-pstn) ;n+1 Zap/g1 available
exten = s,102,Playback(all-circuits-busy-now) n+1
unavailable
exten = s,103,Hangup
Regards
H.G
extension.conf
[sip]
exten = 84,1,Answer
exten = 84,2,Dial(Sip/84,10,t)
exten = 84,3,VoiceMail(u84)
exten = 84,103,VoiceMail(b84)
[fax]
exten = 80,1,Dial(Zap/2,40)
exten = 80,2,Congestion
exten = 80,102,Congestion
[outgoing-pstn]
ingnorepat = 0
exten = _0,1,Dial(Zap/g1/${EXTEN:1})
exten = _0.,1,Dial(Zap/g1/${EXTEN:1})
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[Asterisk-Users] Re: 2 PBX linked via internet
Umair Bari wrote: www.voip-info.org http://www.voip-info.org is one stop ref. for asterisk help. Ok, thanks. I'll go there right away. yes its more then possible. You can put asterisk server at your place and make your friend login to your ast box. you can make your friend a sip account use xlite ( windows based softphone ) at your friends place. IMHO you need to buy fxo card to plug your telco or vonage line into ast box. Ok. I'll decide whether I buy a card or get a VoIP service... yes then your friend can use your vonage or telco line to dialout, and you can make dialplan which actually let your friend chose how to dial and from which line. Cool, super. I'll probably be able to figure out the rest. Thanks a lot! regards, Umair bari On 12/15/05, *Ugo Bellavance* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I think it is the first time I post on this list, I went through the few couple of hundred last messages on the list and couldn't find an answer. I also read the FAQ. I'd need some advice... here is what I plan to do. I've never installed asterisk so I know I need to do some testing. I don't mind buying equipment, but I don't want to overspend, especially in the test phase. I planned on using [EMAIL PROTECTED] on old boxes that I can get for less than 100$ (P3 500/128 MB) or astlinux on pcengines WRAP machines (or even astlinux on the P3). I just bought a WRAP that is going to eventually replace my netgear home router, but I could use the WRAP for testing. There is no hurry to replace the netgear. What I want to do is have one asterisk PBX here and one at my friend's house. We are both with the same ISP, cable modem, with good bandwidth. My friend lives ~ 25 Km from me. - The incoming phone line would be at my house, but I'd like to have an extension going to my friend's. I am testing traffic shaping rules with my firewall right now, so this is likely to be in the plan as well. At home I have 2 lines: one regular telco line and one VoIP line from Vonage (with a motorola phone adapter). The vonage has unlimited long distance USA/Canada (I'm in canada btw). Here are my questions: - Is that possible? Any links to howtos? - Could I connect both of my lines on the pbx? - Can my friend dial through my vonage line to make long-distance calls? Can he actually choose? - I'd like to avoid buying digium cards at least for the testing phase... I know I'll probably need one to use my telco line or the IP line behind the phone adapter, but if I can avoid buying one for my friend's PBX, it would be great. Can I use (free or inexpensive) softphones with asterisk? Thanks in advance, -- http://lists.digium.com/mailman/listinfo/asterisk-users -- Ugo - Please don't send a copy of your reply by e-mail. I read the list. - Please avoid top-posting, long signatures and HTML, and cut the irrelevant parts in your replies. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: 2 PBX linked via internet
Dovid Bender mailto:[EMAIL PROTECTED] wrote: Yes you can do it without a problem. You can set up one at your house and one in his house. The server in your house will require a card. Unless you want to use a VOIP line. You can get an unlimited VOIP line from Broadvoice.com. This will eliminate the need for any ATA's (simmular to your vonage box) and a voice card. I believe they charge $25.00 or $30.00 er month. Ok, but they don't seem very canada-friendly, although canada is covered by their plans... But I know there are companies like that in canada as well. If all you want is for your friend to call out over your lines you do not need to have a PBX on his side. You can have a phone by him (be it a soft phone or hard phone) and have it connect over the internet to your PBX. A hard phone would need to be an IP phone, though, right? There are several software phones out there that are free. You can try xlite. I believe the URL is www.xten.net. Cool! Thanks! Good luck with your project. If you have any questions feel free to ask. Thanks! Regards, Dovid __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: voicemail cutting out
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I have a user that when they go to leave a voicemail message for another user it cuts them off because the talk to quiet. When they talk loud it works fine. Is there a way to fix this In voicemail.conf in general context add line silencetreshold=128 if you put it on 500 you'll have to TALK REALY LOUD :)) -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Wildcard TDM2400P: comments
Question about the TDM2400P. I have a customer that purchased one with a breakout box that has 2 RJ-21 connections male and female and 24 analog jacks. He has a T1 coming into the building which goes to a T1 Router and then from that he has a RJ21 connection going into his old phone system. He wants to remove the connection from his old phone system and plug it into the breakout box and then from the break out box plug into the Digium 2400P. Will that work? On 12/15/05, yusuf [EMAIL PROTECTED] wrote: We are also looking for analog port for fax and dialup modem. Yusuf, would you pls descript what stability issues you had with TDM400P? We are thinking about using TDM400P or Voicetronix OpenPCI-8S. Cheers, Isaac Well, we used the TDM400P a while back, obviously things might have been fixed sinc then. The issues I had were: the card just hanging the machine, or the card just stop taking calls. Can you define a LOT of pots line? Have you considered a channel bank. Here I'm running an ADTRAN 750. It's painless. You just need 1 T1 interface card for 24 lines. Jacques round about 30 pots lines. channel bank might be an option thanks Jacques yusuf wrote: Hi all, we have the need for alot of plain analog lines. We thinking of buying the new Wildcard TDM2400P. Does anybody have any comments with using this card, with any version of Asterisk, (maybe ill make this one Asterisk 1.2.x). I have had some stabilty issues using the 4 TDM400P. What about this new TDM2400P??? thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Phone: 978-203-3848 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ChanIsAvail()
On 12/15/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello,I configure a asterisk server with tdm400p .I wishto set chanisavail() in order to allow users or the
hylafax server to dial numbers to pstn .however Ican't write the rulesto forward requests to the dialpattern when channel is available.I try this however priority 2 fail.how can i forward requests to outgoing-pstncontext ?
exten = s,1,ChanIsAvail(Zap/g1)exten = s,2,Goto(outgoing-pstn) ;n+1 Zap/g1 availableexten = s,102,Playback(all-circuits-busy-now) n+1unavailableexten = s,103,HangupRegards
H.Gextension.conf[sip]exten = 84,1,Answerexten = 84,2,Dial(Sip/84,10,t)exten = 84,3,VoiceMail(u84)exten = 84,103,VoiceMail(b84)[fax]exten = 80,1,Dial(Zap/2,40)
exten = 80,2,Congestionexten = 80,102,Congestion[outgoing-pstn]ingnorepat = 0exten = _0,1,Dial(Zap/g1/${EXTEN:1})exten = _0.,1,Dial(Zap/g1/${EXTEN:1})
It's very important to know what version of asterisk you are using, since as of 1.2 it doesnt do priority jumping.
You'd have to use ChanIsAvail( Zap/g1, j ) if you're using 1.2+
also keep in mind that ${AVAILCHAN} will return something like Zap/2-1 indicating that Zap/2-1 is available in Zap/g1
Another thing is that you're making your incoming calls go to another
context with no idea of what to do there, you should use something like
background to let the users punch in the number they wish to call.
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Re: [Asterisk-Users] RE: Wildcard TDM2400P: comments
Tom Vile wrote: Question about the TDM2400P. I have a customer that purchased one with a breakout box that has 2 RJ-21 connections male and female and 24 analog jacks. He has a T1 coming into the building which goes to a T1 Router and then from that he has a RJ21 connection going into his old phone system. He wants to remove the connection from his old phone system and plug it into the breakout box and then from the break out box plug into the Digium 2400P. Will that work? Why would the breakout box be necessary, just for a connector gender change or something? He could just connect the RJ21 cable from the channel bank directly to the TDM2400P. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hint on Zap channels
this does work, and is adding the hint to the channel on the isdn-card and you can also add a watch to the second-isdn-channel of the card is it possible to use the cid of a isdn-phone as well to identify multiple devices behind one line ? [EMAIL PROTECTED] wrote on 15.12.2005 14:31:09: [EMAIL PROTECTED] ha scritto: has anyone an working example of a hint-entry with a Zap-Channel ? I've got hint working with SIP and SCCP but Zap doesn't seem to work Fixed in current CVS 1.2 and HEAD older versions have a case sensitivity issue so you have to write it in the right way this one works exten = 1, hint, Zap/1 this one does not work exten = 1, hint, ZAP/1 this one does not work exten = 1, hint, zap/1 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ChanIsAvail()
I do not think that Chanisavail will work with a group...
If is does you still need to add the j option to it so that it will Jump
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, December 15, 2005 9:14 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] ChanIsAvail()
Hello,
I configure a asterisk server with tdm400p . I wish to set
chanisavail() in order to allow users or the hylafax server
to dial numbers to pstn . however I can't write the rules
to forward requests to the dial pattern when channel is available.
I try this however priority 2 fail.
how can i forward requests to outgoing-pstn context ?
exten = s,1,ChanIsAvail(Zap/g1)
exten = s,2,Goto(outgoing-pstn) ;n+1 Zap/g1 available exten
= s,102,Playback(all-circuits-busy-now) n+1 unavailable
exten = s,103,Hangup
Regards
H.G
extension.conf
[sip]
exten = 84,1,Answer
exten = 84,2,Dial(Sip/84,10,t)
exten = 84,3,VoiceMail(u84)
exten = 84,103,VoiceMail(b84)
[fax]
exten = 80,1,Dial(Zap/2,40)
exten = 80,2,Congestion
exten = 80,102,Congestion
[outgoing-pstn]
ingnorepat = 0
exten = _0,1,Dial(Zap/g1/${EXTEN:1}) exten =
_0.,1,Dial(Zap/g1/${EXTEN:1})
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RE: [Asterisk-Users] Starting RTP with Dial and MusicOnHold
-Original Message- From: Elton Machado [mailto:[EMAIL PROTECTED] Sent: 15 December 2005 14:03 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Starting RTP with Dial and MusicOnHold Why not to use r option in Dial(SIP/xyz,,r) to simulate the ring? Regards, Hi Elton, Tried that one as well. The Dial(,,r) command actually does the opposite of what I want. The r option specifies that no audio, i.e. no RTP stream, should be passed until the call is answered. This option will generate a SIP 180 Ringing response on an incoming call but since in this case the Cerpack switch needs out of band signalling any 180, 183 or other SIP repsonses are ignored for call progress indication. Thanks, Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] function cut()
I thought that app_cut was deprecated in favour of function cut(), but I cant see this in the list or the code as of SVN-trunk-r7472M? Seeing as Ive just edited the dial plan, can anybody shed any light on this, or should I revert back to app_cut? Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] background music...
Someone else may have a different answer, but I think that's an issue with the cell network. I've tried it with my CDMA cellular phone and have seen the same thing. The issue is not with the music. It is the fact that the cell phone codec is not ment to encode music. I have a radio station streaming on my hold music and have called in to listen to the radio station (I have an extension setup that just puts me on hold).. and it will go from music (though sounding under water) to being just static.. and then back again depending on what all is going on. On 12/15/05, Esteban Maestre [EMAIL PROTECTED] wrote: Hi, list! I have a strange problem. I already tried to get help from the lsit, but I hadn't luck enough... This is the scenario: a GSM mobile phone calls to asterisk. After answering, if asterisk plays background music, when the caller speaks, the music becomes unhearable, too noise, it's not possible to distinguish it from noise. I don't know how to handle this. I've tried several codecs por playing the music, but i don't manage... I want the music to be hearable even if the caller speaks... Did anyone have a similar problem? Can anyone help me? thanks -esteban- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Small explanation of txgain rxgain statement please
I was just looking at: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695. html regarding echo canceller tuning, and I noticed the statement Most people find that they need an rxgain level around 8.0 to have good echo cancellation. The txgain setting varies from installation to installation. Which feels a bit wrong :) Could someone explain why increasing the gain on the inbound zap leg (rxgain) would improve echo cancellation? Of have I misunderstood the roles and meanings of rxgain and txgain? If you read that quote verbatim, it's wrong. The rxgain and txgain settings for analog pstn interfaces should be about 2 db less then the pstn cable loss to the central office. However, due to limitations within the asterisk echo canceller, one typically cannot use that objective as it will result in echo. A very valid starting point is to find out what your pstn cable loss really is to the central office. That value can be obtained through using a telephony transmission test set (rather expensive at about $600 new), talking with someone knowledgable in the telco (sometimes very hard to find that person), complaining to the telco about poor transmission levels and watching over the shoulder of the technican when he measures it from your site (assuming the telco dispatches a technician). The transmission loss value will likely be something between -3db and -12db, depending upon the distance between your site and the central office via the cable path. If that value is measured at -8db (as an example only), then a good starting value for rxgain and txgain is roughly 6db. (You are trying to set the gain values to compensate for the cable loss. Trying to set the gain so the end result is no loss (as in 0db) will definitely cause echo. An informal telco approach is to set it for about a 2db total loss. Therefore, a measured loss of -8db with a txgain/rxgain value of +6db results in a total loss of -2db.) If you start with that known value, then the only reasonable way to adjust your analog interface gain is began decreasing those settings by about 2db per attempt, place a call, and listen for the echo. (Preferably place the test call from your system to another analog pstn phone, not a long distance phone or cell phone as they can inject other potential echo issues that will confuse your tests.) Be sure to stop asterisk and restart it after each gain setting change; a simple reload will not recognize the changes. As an example, my asterisk system is about 7db from the central office. My rxgain=5 and txgain=0 result in very acceptable use with only a slight amount of echo during the first few seconds of a call. The audio is still much lower then desired, but very usable. All of the above assumes that your analog pstn interface card is properly set to match the impedence of the line. In the US, that is 600 ohms which happens to be the default installation value for digium analog cards. Anyone that would suggest a specific gain setting for everyone does not understand analog telephony whatsoever. I'd have to guess the person that wrote the referenced material didn't actually intend to imply 8db would work for everyone. A poor man's way to estimate the pstn cable loss is to use an el-cheapo multimeter, set the multimeter to measure current (milliamps), and place the probes directly across tip ring of the phone line with nothing else attached to the pstn line. You should read something between 20 and 60 milliamps. Send that measured value to me (off list) and I'll convert it to a reasonable rxgain setting. (If you understand ohms law and know that 24 gauge pstn cable is 52 ohms per 1,000 feet, central office voltage is 48 volts, and 24 gauge pstn cable has a loss of 2.31 db per 1,000 feet, then you can compute it yourself. Example: if you measure 26 milliamps, the ohms law calculation indicates you are 1,846 ohms from the CO. 1,846 ohms minus 600 ohms (for CO equipment) divided by 52 ohms indicates you are about 23,960 feet from the CO. 23.9 kilofeet divided by 2.31 db loss per 1,000 feet indicates you can expect a loss of about -10.3db. Then start with an rxgain setting of 8db.) If you are going to sell asterisk pbx's to your customers, then invest in a transmission test set and get the local telephone number for the milliwatt generator. It will be substantially more accurate then the poor man's method shown above. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] astcc issue
Hi list: I need to create a routes list to specific card number wih different prices than the initial routes list ,because markup donot achieve my purpose and markup use for changing prices for all routes,and i need to change prices for specific routes. So is there any possible way to do that? Regards; jonny __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to disable sip Native bridge
Hi, Here is what I have in my sip.con [148] type=friend username=148 secret=something host=dynamic canreinvite=no qualify=no context=interne dtmfmode=rfc2833 mailbox=148 language=fr [spa3kphone00] type=friend host=dynamic context=interne secret=something dtmfmode=rfc2833 disallow=all allow=ulaw canreinvite=no language=fr The spa3000 is on 10.10.11.10 and the phone (Mitel 5215) is on 10.10.10.52. The spa3k can send packets to the Mitel, but not the other way. I would like this this setup to remain the same The asterisk has one nic on each network 10.10.11.1 and 10.10.10.25. The asterisk box is configured to do the ip forwarding for the sipura (the gateway) so that some pc can access the configuration panel of the sipura but the mitel does'nt have a route to the sipura. Only the FXS port is used on the SPA3K with a phone. The problem is when there is a call from or to the spa3k, asterisk try to do a Native Bridge and fail to do so. After when I hangup the FXS on the SPA3K, Asterisk do not get the end of the call. On the other side, if I hangup the Mitel, everything works ok. If I hang-up the call from the sipura or the Mitel before asterisk try the native bridge, everything is ok. Thanks a lot for your help. ___ Jean-François Rousseau www.sys-tech.net [EMAIL PROTECTED] Tél. 24h (418) 520-0739Télec. (418) 520-4554 1-877-969-tech Ouverture Technologique -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Steve Totaro Envoyé : 14 décembre 2005 19:57 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: [Asterisk-Users] How to disable sip Native bridge Hi, I'm trying to disable Native bridges between two SIP Phones. This is because they both see the asterisk box, but they can't see eachother (no it isn't because of NAT). I've tried putting canreinvite=no everywhere in my config, but asterisk is still trying a native bridge on the call. The problem is that when this happen, the native bridge fail but one phone (Sipura 2000) think that the bridging was done and the BYE is not received by asterisk when the call end. So the question is, Is there a way to disable this behavior ? Thanks Post your SIP conf. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fax detected, but no fax extension
It's a Zaptel thing. That's why you put faxdetect=yes in Zapata.conf. When
Zaptel hears the fax tone, it shuts off echo cancellation and tells
Asterisk to shunt the call to the fax extension.
If you are using PRI, for reliable fax detection, you should have something
like this in your default inbound context
exten = s,1,Answer()
exten = s,2,Wait(3)
The reason you Wait() for 3 seconds is because of the cadence of the fax
tone. Because Zaptel hardware answers the call and start processing it
almost instantaneously, it starts processing the call as a voice call.
Because the cadence of the tone is something like 2 seconds on, 2 seconds
off, a whole bunch of stuff can happen in your Asterisk box before it hears
the fax tone. If you jump to another context without having a fax extension
in that context, and then Zaptel hears the fax tone, a lot of your fax calls
will fail or an internal extension will ring, then as the user picks up,
Asterisk goes to the fax extension, and the call to the user is terminated.
This is also a double edged sword: We had a realtor who would call us up
while faxing stuff on his fax machine with the speaker on (who DOES that,
anyway?) Asterisk would pick up, do it's wait() and would hear the realtor's
fax machine in the background, and jump to the fax extension. Fun to
troubleshoot.
If you are using analog hardware, I'm not sure the wait() is necessary
because of the relative slowness of the call setup (i.e. it waits anyway for
the Caller ID)
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 15, 2005 3:41 AM
To: Colin Anderson
Cc: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Fax detected, but no fax extension
OK,
Is Asterisk able to switch incoming calls according to
fax or voice to the right extension .
Which function detect incoming signal ?
Regards
H.G
--- Colin Anderson [EMAIL PROTECTED]
a écrit :
You need an extension called fax in your [fax]
context like this:
[fax]
exten = fax,1,Goto(macro-faxreceive,s,1)
hth
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 14, 2005 3:18 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Fax detected, but no fax
extension
Hello,
I get this message when i send fax Fax detected,
but
no fax extension.
I read mailing list .
Can we solve this ?
my conf :
=PSTN==(fxo)-Asterisk-(fxs)==Hylafax==SMTP/POP3
server
zapata.conf:
context=fax
faxdetect=both
signalling=fxo_ks
group=2
channel = 2
extension.conf
[fax]
exten = 80,1,Dial(Zap/2)
ingnorepat = 0
include = outgoing-pstn
[outgoing-pstn]
exten = _0,1,Dial(Zap/g1/${EXTEN:1})
exten = _0.,1,Dial(Zap/g1/${EXTEN:1})
Regards
H.G
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Re: [Asterisk-Users] sharing a line w/multiple extensions
Rich Adamson wrote: The traditional pbx vendors (back then) would always use the same words that Kevin used, emphasizing the differences between key systems and pbx's. However, many of the pbx manufacturers finally realized they were loosing revenue due to those limitations, and began implementing key-system-type functions in their pbx's. They were not trying to address the key system market, but rather make their pbx products more valuable from a user's perspective. Those that are influencing or controlling the direction of asterisk haven't learned that lesson as yet, partially because of the lack of functionality in the sip phones themselves and partially because asterisk is being developed through the open source community (limited development resources and no published long term plan). Those individuals that have worked towards developing the sip rfc standards have recognized some of the key system vs pbx needs, and have added to the sip standards. However, it takes a while for the sip phone manufacturers (and voip pbx manufacturers) to implement those standards, and in some cases, the manufacturers purposefully leave out certain functions in their sip products to protect their investments in proprietary products. It certainly is not difficult to visualize how voip switching products (such as asterisk or any of the commercial products) could be oriented towards being a switch and address the needs of key systems, pbx's, and central office switching in the same basic product. All of the same functions are required in each case. Asterisk will get there, it will just take a little longer since there isn't any published long term plan to influence the short term development. (No offense intended to any asterisk individual or group; just the nature of most open source development.) I can also assure you that several large companies (most of those company names likely wouldn't be recognized by many of the readers here) are watching the asterisk development closely, and likely are in fear of various open source products negatively impacting their core business. They will adjust their product development (and plans) in an effort to remain one (or more) steps ahead from a marketing sales perspective. Thanks for the history on PBX and key systems. History has a way of repeating itself. I think Asterisk will have to implement features of a key system in the near future. Just judging from the reaction from friends and family who are fascinated by my Asterisk installation, there is huge demand for this kind of system. Digium is just losing out on sales. Is there an open source key system? What other alternative systems are there? How about OpenPBX? Are they integrating any key system support? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1 Echo (was: Small explanation of txgain rxgain statement please)
On 12/15/05, Rich Adamson [EMAIL PROTECTED] wrote: I was just looking at: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695. html regarding echo canceller tuning, and I noticed the statement Most people find that they need an rxgain level around 8.0 to have good echo cancellation. The txgain setting varies from installation to installation. Which feels a bit wrong :) Could someone explain why increasing the gain on the inbound zap leg (rxgain) would improve echo cancellation? Of have I misunderstood the roles and meanings of rxgain and txgain? [snip] Many thanks for clearing that up for me :) the largest part of my misunderstanding was caused by not noticing that that article was referring to the tuning of an FXO line. I am in fact trying to find information on the tuning of an E1 to reduce echo. (Doh!) In theory of course an E1 should work with rxgain=0.0, txgain=0.0 (assuming there is no digital messing going on in the network) and the echo canceller should have a relatively easy job of cancelling echo given that the large majority of the UK phone network is digital, and only the last leg at the far end is usually analogue. I am running Asterisk 1.0.9, and have backported the KB1 canceller into Zaptel 1.0.9.2, which does not seem to have caused any problems. Nor has it really caused any improvement though :) I am beginning to wonder whether what echo IS heard is being caused by packetisation delays in the network - The default tap length is 128, or I believe 16ms. If something in the PSTN causes a delay more than that length (no idea what might cause that) then echo would still be heard. Does anyone have any experience in this area? Any ideas? How heavy handed would it be to increase the tap length to 256? I have not seen anyone suggest that this might be a good idea. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pops in Call Recordings Tied to Dropped Audio in Calls
Kevin P. Fleming wrote: Matt Roth wrote: We have no hardware timing device on the box (no Zap hardware) and are using the 2.6 kernel as the timing source. Digium tech support told us this is better than ztdummy, which we were using before. We experienced the same problems then, as well. Could a lack of a hardware timing source be our problem? If you don't have ztdummy loaded, you don't have a timing source. Asterisk does not take timing from the 'kernel', although ztdummy has a mode where it can do so. That advice about not loading ztdummy came from our paid support, including the bit about Asterisk falling back to the kernel for timing if no other source is available. It's concerning to me, to say the least, that we are paying for misinformation. We are now running Asterisk in a production environment with no timing source. We can't change this until the end of the business day. Luckily, only our MOH *should* be affected, but we are trying to troubleshoot problems and introducing any source of strange behavior makes that more difficult. We also wasted time last night testing whether or not removing ztdummy solved our problem. Paid support time (since we were on call with Digium while doing the testing) as well as time that could've been used to look into real solutions. I'm not asking anyone to fix the problem for us, but I would like some legitimate feedback on its possible sources. From the list, I consider it a gift. From paid tech support, I consider it a responsibility. Does ztdummy fall to the kernel mode by default, or does it have to be configured to do so? Regardless, if the call is SIP to SIP, a timing source is not relevant for this problem. Noted. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pops in Call Recordings Tied to Dropped Audio in Calls
Matt Roth wrote: That advice about not loading ztdummy came from our paid support, including the bit about Asterisk falling back to the kernel for timing if no other source is available. It's concerning to me, to say the least, that we are paying for misinformation. We are now running Asterisk in a production environment with no timing source. We can't change this until the end of the business day. Luckily, only our MOH *should* be affected, but we are trying to troubleshoot problems and introducing any source of strange behavior makes that more difficult. Please email your ticket number to me directly (_not_ to the list). We also wasted time last night testing whether or not removing ztdummy solved our problem. Paid support time (since we were on call with Digium while doing the testing) as well as time that could've been used to look into real solutions. I'm not asking anyone to fix the problem for us, but I would like some legitimate feedback on its possible sources. From the list, I consider it a gift. From paid tech support, I consider it a responsibility. Agreed 100%. Does ztdummy fall to the kernel mode by default, or does it have to be configured to do so? On 2.6 kernels, ztdummy can use either the kernel ticks (jiffies) for timing, or the hardware realtime clock (RTC). By default it uses the RTC when built against kernel headers for 2.6.13 or newer; it can be manually configured to use the RTC for older kernels. The RTC is a more reliable timing source than kernel ticks, because they can be delayed small amounts due to the kernel performing process switches and the like, although usually there is not a great deal of difference. Also, on recent 2.6 kernels the jiffy frequency is adjustable at configuration time and no longer defaults to 1000 per second; if RTC mode is not used, then the kernel frequency _must_ be 1000Hz for ztdummy to work correctly. On 2.4 kernels the situation is very different: ztdummy on 2.4 kernels uses a USB device for timing generation. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] function cut()
Ok, looks like app_cut is also function cut, but is missing from the Makefile in apps? If you add it in it doesnt compile, due to cut_synopsis not being defined. Seems like its in a state of flux, whos working on this? Steve From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 15 December 2005 14:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] function cut() I thought that app_cut was deprecated in favour of function cut(), but I cant see this in the list or the code as of SVN-trunk-r7472M? Seeing as Ive just edited the dial plan, can anybody shed any light on this, or should I revert back to app_cut? Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sharing a line w/multiple extensions
The traditional pbx vendors (back then) would always use the same words that Kevin used, emphasizing the differences between key systems and pbx's. However, many of the pbx manufacturers finally realized they were loosing revenue due to those limitations, and began implementing key-system-type functions in their pbx's. They were not trying to address the key system market, but rather make their pbx products more valuable from a user's perspective. Those that are influencing or controlling the direction of asterisk haven't learned that lesson as yet, partially because of the lack of functionality in the sip phones themselves and partially because asterisk is being developed through the open source community (limited development resources and no published long term plan). Those individuals that have worked towards developing the sip rfc standards have recognized some of the key system vs pbx needs, and have added to the sip standards. However, it takes a while for the sip phone manufacturers (and voip pbx manufacturers) to implement those standards, and in some cases, the manufacturers purposefully leave out certain functions in their sip products to protect their investments in proprietary products. It certainly is not difficult to visualize how voip switching products (such as asterisk or any of the commercial products) could be oriented towards being a switch and address the needs of key systems, pbx's, and central office switching in the same basic product. All of the same functions are required in each case. Asterisk will get there, it will just take a little longer since there isn't any published long term plan to influence the short term development. (No offense intended to any asterisk individual or group; just the nature of most open source development.) I can also assure you that several large companies (most of those company names likely wouldn't be recognized by many of the readers here) are watching the asterisk development closely, and likely are in fear of various open source products negatively impacting their core business. They will adjust their product development (and plans) in an effort to remain one (or more) steps ahead from a marketing sales perspective. Thanks for the history on PBX and key systems. History has a way of repeating itself. I think Asterisk will have to implement features of a key system in the near future. Just judging from the reaction from friends and family who are fascinated by my Asterisk installation, there is huge demand for this kind of system. Digium is just losing out on sales. Doubt they are losing much in sales. Sales of the digium cards are from geeks (like many of us on this list) and small companies selling asterisk into business accounts. Most 'friends family' wouldn't consider investing $1k for all the pieces necessary to have a reasonable system (even if they could use a retired PC) unless they're geeks as well. Is there an open source key system? What other alternative systems are there? How about OpenPBX? Are they integrating any key system support? Not that I'm aware of, but I don't try to keep track of competing projects either. There's sort of a dichotomy thing going on where most development folks (whether its asterisk or some other I/T-type projects) are focused on programming some function/features that are system oriented (eg, odbc, sql support, echo cancellers, fax support, menues, jitterbuffers, architectural changes, scipts); and, another group without programming skills that would love to see additional basic pbx/key-system functions implemented that don't require someone to jump through hoops in the dialplan. But, that's the nature of open source projects. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outbound Routing
Hello, I have a 4 port FXO digium card with 3 PSTNs attached to it and AsteriskAtHome setup. Everything is working fine except outbound calls. When I dial a outside number, it works fine, but when another employee trys to dial out while I am on a line, it will not go. I have a outgoing route setup in the AMP interface. Dial Pattern: 1NXXNXX NXXNXX NXX Trunk Sequence: ZAP/3 ZAP/2 ZAP/1 Any ideas why when someone is on ZAP/3 and someone else trys to call out, it does not pick the next available PSTN? Thanks, Brendan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1 Echo (was: Small explanation of txgain rxgain statement please)
I was just looking at: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695. html regarding echo canceller tuning, and I noticed the statement Most people find that they need an rxgain level around 8.0 to have good echo cancellation. The txgain setting varies from installation to installation. Which feels a bit wrong :) Could someone explain why increasing the gain on the inbound zap leg (rxgain) would improve echo cancellation? Of have I misunderstood the roles and meanings of rxgain and txgain? [snip] Many thanks for clearing that up for me :) the largest part of my misunderstanding was caused by not noticing that that article was referring to the tuning of an FXO line. I am in fact trying to find information on the tuning of an E1 to reduce echo. (Doh!) In theory of course an E1 should work with rxgain=0.0, txgain=0.0 (assuming there is no digital messing going on in the network) and the echo canceller should have a relatively easy job of cancelling echo given that the large majority of the UK phone network is digital, and only the last leg at the far end is usually analogue. That last leg is usually part of the problem since there is going to be a hybrid conversion. I am running Asterisk 1.0.9, and have backported the KB1 canceller into Zaptel 1.0.9.2, which does not seem to have caused any problems. Nor has it really caused any improvement though :) The KB1 canceller improves echo, but it appears as though it achieved better results by forcing half-duplex communications. From a pure non-technical user perspective, the quality of a telephone conversation has been lowered simply because humans are use to communicating in full duplex mode. I am beginning to wonder whether what echo IS heard is being caused by packetisation delays in the network - The default tap length is 128, or I believe 16ms. If something in the PSTN causes a delay more than that length (no idea what might cause that) then echo would still be heard. Certainly not hard to change the tap length and eval it. Does anyone have any experience in this area? Any ideas? How heavy handed would it be to increase the tap length to 256? I have not seen anyone suggest that this might be a good idea. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] function cut()
Added it to the Makefile, amended the reference to cut_synopsis and it compiles, installs and works fine. Logged it as a bug on mantis. Steve From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 15 December 2005 16:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] function cut() Ok, looks like app_cut is also function cut, but is missing from the Makefile in apps? If you add it in it doesnt compile, due to cut_synopsis not being defined. Seems like its in a state of flux, whos working on this? Steve From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 15 December 2005 14:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] function cut() I thought that app_cut was deprecated in favour of function cut(), but I cant see this in the list or the code as of SVN-trunk-r7472M? Seeing as Ive just edited the dial plan, can anybody shed any light on this, or should I revert back to app_cut? Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] traffic shaping
On Wednesday 14 December 2005 12:58, Colin Anderson wrote: http://www.krisk.org/astlinux/misc/astshape Nice, it's a cleaned up version of my script. Too bad they didn't credit it properly. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sharing a line w/multiple extensions
Fascinating discussion. The whole idea of acceptance of an asterisk based system by the rest of the family is probably worthy of its own thread. I'm in alpha test (I switch on asterisk after the wife leaves for work, switch it back before she gets home ;-) ) of my home asterisk system, so I've been thinking/worrying about a lot of similar issues. I'm particularly worried about acceptance of this shared line (or lack thereof) aspect of the system. My wife will get the idea of extensions, transfers, parking, etc. because she uses a PBX at work, though I worry that the habits of how the phone is supposed to work at home may die hard with her. And the kids are a whole 'nuther story. I thought that having some common area phones share a single extension (wired into a single ATA FXS port) might ease the transition, but I'm also afraid it might be confusing (you can just pick up from these extensions, but you have to transfer or park to/from these extensions. Huh?). The huge selling point, which I'm hoping will overcome any initial resistance, is the idea that one person will no longer tie up the whole phone system for the house when they make/take a call. And deploying one of my free DIDs to give my 16-year-old his own phone number that rings only in his bedroom is the real ace up my sleeve! Sure, Asterisk will come with a lot of other neat features, but frankly most of them have more geek appeal (though I have high hopes for my favorite feature -- announced caller id over the stereo/tivo while we're making dinner -- to revolutionize the way we deal with (or at least who answers ;-) ) phone calls at that hour), and in some cases I think may face similar that's not the way it's supposed to work objections. For example, while they will acknowledge that voicemail is cool, I suspect they'll miss the simplicity of walking into the kitchen, seeing if the answering machine is blinking, and just pressing the button. I'm excited AND anxious about starting a real beta test with them! Maybe that's why I'm already 3 weeks behind my original schedule. ;-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pops in Call Recordings Tied to Dropped Audio in Calls
Kevin P. Fleming wrote: On 2.6 kernels, ztdummy can use either the kernel ticks (jiffies) for timing, or the hardware realtime clock (RTC). By default it uses the RTC when built against kernel headers for 2.6.13 or newer; it can be manually configured to use the RTC for older kernels. The RTC is a more reliable timing source than kernel ticks, because they can be delayed small amounts due to the kernel performing process switches and the like, although usually there is not a great deal of difference. Also, on recent 2.6 kernels the jiffy frequency is adjustable at configuration time and no longer defaults to 1000 per second; if RTC mode is not used, then the kernel frequency _must_ be 1000Hz for ztdummy to work correctly. We're running a 2.6.12 kernel (Murphy's Law!), so I could use some tips for manually configuring ztdummy for RTC. We're running ABE (A.2-1), but that shouldn't be a problem since all of the zaptel related stuff is still built from source. Our kernel is a stock Fedora Core 3 kernel (2.6.12-1.1376_FC3smp), so the frequency is most likely not set at 1000Hz. Is there an easy way to verify this? Would we be better off buying a Digium card to use solely as a timing source, or is ztdummy adequate? We tried putting an X100P in the box, but the BIOS didn't recognize it. We have an extra quad-span card, but it's 5-volt and our PCI slots are 3.3 (Murphy's Law!). I was under the impression that timing was only an issue for MOH, IAX trunking, and MeetMe conferencing, then I found this while doing some research on configuring ztdummy for RTC: - [Asterisk-Users] Popping and Clicking on Local WAN with X-Lite (http://lists.digium.com/pipermail/asterisk-users/2004-October/065807.html) It sounds curiously similar to our issue, but I've pointed out some areas where we're unique below. Could this be the source of our problem? - We're using a Cisco AS5400 Universal Gateway to terminate our Ts (it forwards all calls to Asterisk via SIP), so IRQ sharing shouldn't be a problem (it's also why we have no Zap hardware in the box). Things look pretty good in /proc/interrupts as well. [EMAIL PROTECTED] ~]# cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 0: 10513090 15234131 15354631 15354147IO-APIC-edge timer 1: 2 1 6 4IO-APIC-edge i8042 8: 33 48 40 68IO-APIC-edge rtc 9: 0 0 0 0 IO-APIC-level acpi 12: 22 25 25 27IO-APIC-edge i8042 14: 6566 71549 5166 29898IO-APIC-edge ide0 50: 30 9369 424384 159692 IO-APIC-level ide2 58: 46301 29337 41320 43009 IO-APIC-level megaraid 82: 25736221 3 4 3 IO-APIC-level eth2 90: 2 314496 2 3 IO-APIC-level eth3 98: 5 5 7 5 IO-APIC-level ehci_hcd:usb1 106: 13 15 13 17 IO-APIC-level uhci_hcd:usb2 114: 0 0 0 0 IO-APIC-level uhci_hcd:usb3 122:120 3319582 2536 IO-APIC-level uhci_hcd:usb4 NMI: 1305674737642 LOC: 56440137 56450200 56450713 56450888 ERR: 3 MIS: 0 - Some of the information I read pointed out that RTC must be loaded as a module, and not compiled directly into the kernel. I'm unsure if this is still the case, and I'm also not sure how to check whether it's compiled in or not. lsmod and /proc/modules/ don't show it, but it is in /proc/interrupts as shown above. [EMAIL PROTECTED] ~]# lsmod Module Size Used by ztdummy 8800 0 zaptel204328 79 ztdummy crc_ccitt 6593 1 zaptel md5 9025 1 ipv6 306305 22 parport_pc 35497 0 lp 19473 0 parport47693 2 parport_pc,lp autofs427081 0 i2c_dev16577 0 i2c_core 29889 1 i2c_dev nfs 223729 2 lockd 77297 2 nfs sunrpc166457 3 nfs,lockd pcmcia 35925 0 yenta_socket 28617 0 rsrc_nonstatic 17857 1 yenta_socket pcmcia_core60109 3 pcmcia,yenta_socket,rsrc_nonstatic dm_mod 68257 0 joydev 15553 0 video 23241 0 button 9025 0 battery15305 0 ac 9929 0 uhci_hcd 39265 0 ehci_hcd 41037 0 e1000 119341 0 tg3 103109 0 floppy 72857 0 sg 46969 0 ext3 148049 5 jbd71153 1 ext3 megaraid_mbox 44241 6 megaraid_mm17273 1
[Asterisk-Users] Shutting down Asterisk when not in RTP Stream
I'm very confused about something. I have two phones that have reinvited and have an RTP session open. I confirmed this by running ngrep on the Asterisk box. Asterisk still shows the calls on the console. *CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 192.168.10.125 a00090201 45dfabad1bd 00103/0 ulaw No Tx: ACK 192.168.10.4 a00090101 ca3279d8-3e 00102/1 ulaw No Tx: ACK When I shut asterisk down, the call terminates. I don't understand that. If Asterisk isn't in the RTP path, how can shutting it down terminate an active call? Don't know if it's relevant, but the 192.168.10.4 is an OpenSER box. Thanks. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Handyton 486 Outbound problem
I've got a Handytone 486 ATA. It's registering fine with SIP and calls other 2 digit internal extensions just fine. When I try to dial out though (7/10-digit calls), I get a busy signal.How should I troubleshoot this? -- Craig Bruenderman ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Shutting down Asterisk when not in RTP Stream
G! Asterisk sends a BYE to the phone when it gets shut down. What a pain. Eventhough it isn't in the RTP path, it must keep track of it's current call state, and when you shut it down, terminate all those calls. Reason I am trying this is that I've had asterisk core dump on me a few times, and I'd like to be able to restart it without losing calls in progress. Doug. -Original Message- From: Douglas Garstang Sent: Thursday, December 15, 2005 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Shutting down Asterisk when not in RTP Stream I'm very confused about something. I have two phones that have reinvited and have an RTP session open. I confirmed this by running ngrep on the Asterisk box. Asterisk still shows the calls on the console. *CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 192.168.10.125 a00090201 45dfabad1bd 00103/0 ulaw No Tx: ACK 192.168.10.4 a00090101 ca3279d8-3e 00102/1 ulaw No Tx: ACK When I shut asterisk down, the call terminates. I don't understand that. If Asterisk isn't in the RTP path, how can shutting it down terminate an active call? Don't know if it's relevant, but the 192.168.10.4 is an OpenSER box. Thanks. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan-capi avm b1 and capi.conf problems
On Thu, Dec 15, 2005 at 02:09:59PM +0100, Ricardo wrote: The main problem is that can load asterisk (asterisk -vvvcg) but i get just one error message and, off course isdn do not work: Dec 15 11:13:25 ERROR[4887]: chan_capi.c:4835 load_module: Unable to load config capi.conf, CAPI disabled You need /etc/isdn/capi.conf and /etc/asterisk/capi.conf. The second is used by the asterisk module chan_capi. Check file permissions and content of /etc/asterisk/capi.conf. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voipsupply - my experience
Hi, I would just like to let everyone know about the support I have rec'd from voipsupply. I travelled from .au to the US to do some urgent server upgrades for $EMPLOYER. During this trip I has impressed upon me (with absolutely zero notice) the requirement for a IP Phone link to the .au office. With only a couple of day left in the US I was stumped. I called Voipsupply and left a message (it was 9:30PM CST - Houston) at 10:30PM Mark emailed me and sent me his Cell phone number. The upshot is: I placed an order for some Digium kit at Midnight, on the 14th and I have just rec'd the goods in Houston now. Mark: Fantastic service! I appreciate the help. Regards, -- Terry Gilsenan Information Systems Manager InterOil Corporation ph: +61-7-4046-4614 mb: +61-417-600-360 ===[Disclaimer]=== This electronic transmission, including any attachments, is confidential, may contain privileged information and should be read or retained only by the intended recipient. If you received this message in error, please delete it from your system and notify the sender immediately. Any review, dissemination or other use of this information by persons or entities other than the intended recipient is strictly prohibited. ===[End]=== ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] E1 Echo (was: Small explanation of txgain rx gain statement please)
Does anyone have any experience in this area? Any ideas? How heavy handed would it be to increase the tap length to 256? I have not seen anyone suggest that this might be a good idea. On my PRI, 256 made things bad, super echo-y. Moving back to 128 works good 99% of the time, for me. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Shutting down Asterisk when not in RTP Stream
Do you have 't' or 'T' in the Dial Application? Diyanat From: Douglas Garstang [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [Asterisk-Users] Shutting down Asterisk when not in RTP Stream Date: Thu, 15 Dec 2005 10:38:22 -0700 MIME-Version: 1.0 I'm very confused about something. I have two phones that have reinvited and have an RTP session open. I confirmed this by running ngrep on the Asterisk box. Asterisk still shows the calls on the console. *CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 192.168.10.125 a00090201 45dfabad1bd 00103/0 ulaw No Tx: ACK 192.168.10.4 a00090101 ca3279d8-3e 00102/1 ulaw No Tx: ACK When I shut asterisk down, the call terminates. I don't understand that. If Asterisk isn't in the RTP path, how can shutting it down terminate an active call? Don't know if it's relevant, but the 192.168.10.4 is an OpenSER box. Thanks. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Originating calls to a channel groups
Hi! I'm using the asterisk manager to create calls using the originate action. Can I know what channel selects Asterisk when I originate a call with a channel group? Thank you! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Google Analytics and voip-info.org
Damned! What is going on with voip-info.org this week? I think Google Analytics is the cause... Has anybody facing this problem too? Denis. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to change the Dial command H option to ## ?
Quoting Matt Riddell [EMAIL PROTECTED]: That is a part of Asterisk I am not yet familiar with. I will give it a try Thanks Obelix wrote: I want to use '##' to terminate a call instead of the '*' used by the Dial command's H option. Is there a way to change the key or use another option to achieve the same effect? Application map in features.conf assigning ## to Hangup() ? Maybe :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: HELP!
Just to recap this, I'm now having this problem also. The only difference being, Asterisk is always returning a 401 unauthorized back to the UA and thus the registration never completes. Nothing I do fixes it, what's worse is other then watching debug (and sip show peer blah, naturally) reflects absolutely no errors. -- snippet of debug -- Dec 15 12:17:46 NOTICE[26076]: chan_sip.c:6275 check_auth: stale nonce received from 'sip:[EMAIL PROTECTED]' Transmitting (NAT) to 66.166.222.59:5060: SIP/2.0 401 Unauthorized .. Snip .. Would this suggest a UA problem? Did the obvious, and rebooted it. Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian C. Fertig Sent: Tuesday, October 25, 2005 3:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] HELP! How do I resolve this? -- Unregistered SIP '107' -- Registered SIP '107' at 192.168.0.161 port 5060 expires 60 -- Registered SIP '107*' at 192.168.0.161 port 5066 expires 60 Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' This happens and all of my phones loose registration. Its driving me nuts. ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office 813.864.3161x107 Office 813.864.3164 Direct 813.817.9961 Cellular 813.881.9762 Fax Web: www.planet-telecom.com email: [EMAIL PROTECTED] --IM's--- AIM: planetTelNOC ICQ: 65075522 MSN: [EMAIL PROTECTED] 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Realtime connection failed
Hello. I have been trying to setup Asterisk with the realtime MySql with sip users descibed at http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip I have made a Mysql database called "asterisk" and a table inside called "sip_buddies" as descibed at the wiki page. The Mysqluser "root" hasall rights to the "asterisk" database. I have edited the Extconfig.conf to this: sippeers = mysql,asterisk,sip_buddies and in res_mysql.conf looks like this: [general] dbhost = localhost dbname = asterisk dbuser = root dbpass = passwd When I reload Asterisk, I get this error: ERROR[8280]: res_config_mysql.c:615 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on localhost. Check debug for more info. The Cdr with mysql works fine, and it also connects using the MySql user "root", so it should not be something wring with the Mysql part. Does anyone see any obvious error I've made ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip behind the NAT
On Tue, Dec 13, 2005 at 11:32:15PM -0500, Tom Rymes wrote: On Dec 13, 2005, at 8:25 AM, Michael George wrote: I have a similar problem with a client's system. They have * 1.0.x behind a NAT with all the SIP phones on the local network. Their VoIP provider is outside the NAT (a Metaswitch at their ISP, connected to the phone lines from there). Their network guy has the firewall passing traffic on ports 5060 and 1-2 to the * system. I have externalIP and localnet set, but nat=no (default) is the case for this one. Occasionally they will place outgoing calls and the other party does not hear anything. Usually another attempt at the call will pass audio normally. One person who makes about 100 calls a day remembers having this happen on about 7 calls one day. No one recalls this ever happening on incoming calls (though this client primarily makes outgoing calls, I believe). Apparently this has been happening for a while and they just now mentioned it to me. Would nat=yes in the general section of sip.conf make a difference in this case? Is there anything else I could look at that might alleviate this problem? Without being a smartass, the only way to find out is to see if it works. More obviously, if the Asterisk server has a NAT between it and the ITSP, then use nat=yes, if it doesn't, then use nat=no. Of course, if you set nat=no, then don't bother setting localnet or externip, either. Also keep in mind that some routers' DMZ settings still leave your box behind NAT. They just forward all of the ports to the specified address. (Linksys routers do this.) I didn't detect any smartassity in your response... I'm going to try nat=yes in the general section and then I'm going to trim down the RTP port range just for fun and see what happens. Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] traffic shaping
Andrew Kohlsmith wrote: On Wednesday 14 December 2005 12:58, Colin Anderson wrote: http://www.krisk.org/astlinux/misc/astshape Nice, it's a cleaned up version of my script. Too bad they didn't credit it properly. Looks like an incarnation of WonderShaper to me :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voipsupply - my experience
On Thursday 15 December 2005 11:55, Terry H. Gilsenan wrote: Hi, I would just like to let everyone know about the support I have rec'd from voipsupply. I travelled from .au to the US to do some urgent server upgrades for $EMPLOYER. During this trip I has impressed upon me (with absolutely zero notice) the requirement for a IP Phone link to the .au office. With only a couple of day left in the US I was stumped. I called Voipsupply and left a message (it was 9:30PM CST - Houston) at 10:30PM Mark emailed me and sent me his Cell phone number. The upshot is: I placed an order for some Digium kit at Midnight, on the 14th and I have just rec'd the goods in Houston now. Mark: Fantastic service! I appreciate the help. Regards, Having moved from AU to US last year I know how important a good reliable and cheap link back home is. for this i use digium equipment and asterisk with an AU iax provider. i have a local number in Brisbane. I have also purchased some VoIP equipment from Voipsupply for my employer and have received fantastic service and support from them for that also. Dennis ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Shutting down Asterisk when not in RTP Stream
Nope. If I did, then the phones wouldn't reinvite. -Original Message- From: Diyanat Ali [mailto:[EMAIL PROTECTED] Sent: Thursday, December 15, 2005 11:14 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Shutting down Asterisk when not in RTP Stream Do you have 't' or 'T' in the Dial Application? Diyanat From: Douglas Garstang [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [Asterisk-Users] Shutting down Asterisk when not in RTP Stream Date: Thu, 15 Dec 2005 10:38:22 -0700 MIME-Version: 1.0 I'm very confused about something. I have two phones that have reinvited and have an RTP session open. I confirmed this by running ngrep on the Asterisk box. Asterisk still shows the calls on the console. *CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 192.168.10.125 a00090201 45dfabad1bd 00103/0 ulaw No Tx: ACK 192.168.10.4 a00090101 ca3279d8-3e 00102/1 ulaw No Tx: ACK When I shut asterisk down, the call terminates. I don't understand that. If Asterisk isn't in the RTP path, how can shutting it down terminate an active call? Don't know if it's relevant, but the 192.168.10.4 is an OpenSER box. Thanks. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Google Analytics and voip-info.org
What problem? On 12/15/05, Denis Galvão - iSolve [EMAIL PROTECTED] wrote: Damned!What is going on with voip-info.org this week?I think Google Analytics is the cause...Has anybody facing this problem too?Denis. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Disposition Failed still happening
Is anyone else still having disposition failed showing up in the cdr's on 1.2.1? I can't seem to figure out why asterisk would put that in the cdr's when the calls have in fact completed successfully 0.o Aaron Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Google Analytics and voip-info.org
I think Google Analytics is the cause... You misspelled wiki J ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] traffic shaping
Matt Riddell wrote: Andrew Kohlsmith wrote: On Wednesday 14 December 2005 12:58, Colin Anderson wrote: http://www.krisk.org/astlinux/misc/astshape Nice, it's a cleaned up version of my script. Too bad they didn't credit it properly. Looks like an incarnation of WonderShaper to me :) Doh, just as I sent that I noticed it said based on WonderShaper in the comments and tried to cancel the send :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo Canceller usage
Using a TDM400P with an FXO module and an FXS module, and a zapata.conf with echocancel=yes above both channel definitions, is echo cancelling applied individually to each module when a call is made out to the PSTN? Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voipsupply - my experience
I also have nothing but wonderful things to say about voipsupply.com, they are first on my list for equipment needs. They have great methods of communication and they free up my time to better serve my clients. Omar A. Sabek On 12/15/05, Dennis Gilmore [EMAIL PROTECTED] wrote: On Thursday 15 December 2005 11:55, Terry H. Gilsenan wrote: Hi, I would just like to let everyone know about the support I have rec'd from voipsupply. I travelled from .au to the US to do some urgent server upgrades for $EMPLOYER. During this trip I has impressed upon me (with absolutely zero notice) the requirement for a IP Phone link to the .au office. With only a couple of day left in the US I was stumped. I called Voipsupply and left a message (it was 9:30PM CST - Houston) at 10:30PM Mark emailed me and sent me his Cell phone number. The upshot is: I placed an order for some Digium kit at Midnight, on the 14th and I have just rec'd the goods in Houston now. Mark: Fantastic service! I appreciate the help. Regards, Having moved from AU to US last year I know how important a good reliable and cheap link back home is. for this i use digium equipment and asterisk with an AU iax provider. i have a local number in Brisbane. I have also purchased some VoIP equipment from Voipsupply for my employer and have received fantastic service and support from them for that also. Dennis ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo Canceller usage
Richard Scobie wrote: Using a TDM400P with an FXO module and an FXS module, and a zapata.conf with echocancel=yes above both channel definitions, is echo cancelling applied individually to each module when a call is made out to the PSTN? Individually? Yes... but I don't know how else you are thinking it would be applied. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disposition Failed still happening
I actually opened a bug report on this earlier this month: http://bugs.digium.com/view.php?id=5918 I have tried a new SVN version from a few days ago and it still showed as FAILED for me in the following scenario: incoming call from PSTN ---SIP--- asterisk ---IAX2--- asterisk ---SIP--- SIP Phone At least now it appears that the billsec field is no longer showing zero, but the FAILED disposition is annoying. If I was a programmer I would happily jump in and see what could be done. Maybe in my free time I can squeeze in a lesson in C sometime ;) On 12/15/05, Aaron Daniel [EMAIL PROTECTED] wrote: Is anyone else still having disposition failed showing up in the cdr'son 1.2.1?I can't seem to figure out why asterisk would put that in thecdr's when the calls have in fact completed successfully 0.oAaron Daniel ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo Canceller usage
Kevin P. Fleming wrote: Individually? Yes... but I don't know how else you are thinking it would be applied. Apologies for breaking the thread. Just trying to get an idea of how things work together. I had considered that in this scenario, the echo can on the FXS only has to deal with a tail length back to the FXO hybrid, which on adequate hardware would be so short that any echo would just be sidetone and so could be dispensed with for the sake of CPU usage. The echo can on the FXO would be the one doing the work, on the tail back to the far end hybrid. Or have I misunderstood how Zap EC works? Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disposition Failed still happening
I should note that in the following scenario: incoming call from PSTN ---SIP--- asteriskA ---IAX2--- asteriskB ---SIP--- SIP Phone The call log does show disposition ANSWERED on asteriskA, but FAILED on asteriskB. On 12/15/05, tracinet [EMAIL PROTECTED] wrote: I actually opened a bug report on this earlier this month: http://bugs.digium.com/view.php?id=5918 I have tried a new SVN version from a few days ago and it still showed as FAILED for me in the following scenario: incoming call from PSTN ---SIP--- asterisk ---IAX2--- asterisk ---SIP--- SIP Phone At least now it appears that the billsec field is no longer showing zero, but the FAILED disposition is annoying. If I was a programmer I would happily jump in and see what could be done. Maybe in my free time I can squeeze in a lesson in C sometime ;) On 12/15/05, Aaron Daniel [EMAIL PROTECTED] wrote: Is anyone else still having disposition failed showing up in the cdr'son 1.2.1?I can't seem to figure out why asterisk would put that in thecdr's when the calls have in fact completed successfully 0.oAaron Daniel ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sharing a line w/multiple extensions
Rich, Kevin I just stubbled into one of these projects. I am making it work for the client but constantly running into walls. Client is very cool about the work and loved the history of the Key System from Rich. Current issues and solutions: Busy boxes = SNOM 360s with Sidecars Static Parking(orbit or 60) = Seperate MeetMe rooms (MOH if single user) Paging = Extra SIP line on the phones set to auto answer, ugly chanisavial check No Voicemail = a, bounce to Oper Queues = Hard Code procedural so that I can hint the phones right. answer at any phone = button hack for the blinking button. Maybe I should submit this job to Digium for the contest :-) Andrew On 12/15/05, Rich Adamson [EMAIL PROTECTED] wrote: The traditional pbx vendors (back then) would always use the same words that Kevin used, emphasizing the differences between key systems and pbx's. However, many of the pbx manufacturers finally realized they were loosing revenue due to those limitations, and began implementing key-system-type functions in their pbx's. They were not trying to address the key system market, but rather make their pbx products more valuable from a user's perspective. Those that are influencing or controlling the direction of asterisk haven't learned that lesson as yet, partially because of the lack of functionality in the sip phones themselves and partially because asterisk is being developed through the open source community (limited development resources and no published long term plan). Those individuals that have worked towards developing the sip rfc standards have recognized some of the key system vs pbx needs, and have added to the sip standards. However, it takes a while for the sip phone manufacturers (and voip pbx manufacturers) to implement those standards, and in some cases, the manufacturers purposefully leave out certain functions in their sip products to protect their investments in proprietary products. It certainly is not difficult to visualize how voip switching products (such as asterisk or any of the commercial products) could be oriented towards being a switch and address the needs of key systems, pbx's, and central office switching in the same basic product. All of the same functions are required in each case. Asterisk will get there, it will just take a little longer since there isn't any published long term plan to influence the short term development. (No offense intended to any asterisk individual or group; just the nature of most open source development.) I can also assure you that several large companies (most of those company names likely wouldn't be recognized by many of the readers here) are watching the asterisk development closely, and likely are in fear of various open source products negatively impacting their core business. They will adjust their product development (and plans) in an effort to remain one (or more) steps ahead from a marketing sales perspective. Thanks for the history on PBX and key systems. History has a way of repeating itself. I think Asterisk will have to implement features of a key system in the near future. Just judging from the reaction from friends and family who are fascinated by my Asterisk installation, there is huge demand for this kind of system. Digium is just losing out on sales. Doubt they are losing much in sales. Sales of the digium cards are from geeks (like many of us on this list) and small companies selling asterisk into business accounts. Most 'friends family' wouldn't consider investing $1k for all the pieces necessary to have a reasonable system (even if they could use a retired PC) unless they're geeks as well. Is there an open source key system? What other alternative systems are there? How about OpenPBX? Are they integrating any key system support? Not that I'm aware of, but I don't try to keep track of competing projects either. There's sort of a dichotomy thing going on where most development folks (whether its asterisk or some other I/T-type projects) are focused on programming some function/features that are system oriented (eg, odbc, sql support, echo cancellers, fax support, menues, jitterbuffers, architectural changes, scipts); and, another group without programming skills that would love to see additional basic pbx/key-system functions implemented that don't require someone to jump through hoops in the dialplan. But, that's the nature of open source projects. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger
Re: [Asterisk-Users] Voipsupply - my experience
On Thu, Dec 15, 2005 at 03:03:04PM -0500, Omar A. Sabek wrote: I also have nothing but wonderful things to say about voipsupply.com, they are first on my list for equipment needs. They have great methods of communication and they free up my time to better serve my clients. I'm happy to hear that everyone's had good service from them, but I've felt burned by them twice. The first time, I ordered over $4000 worth of hardware from them and as soon as my order is processed, I get spammed with all their monthly specials (all in caps, of course). I sent them an e-mail asking them why they treat their customers this way, but I never received a response. Also, this order was split up and shipped at different times, but their order status pages didn't explain this (and I had no idea where my parts were for a week). I gave them another chance recently--just last week, in fact. I ordered two IAXy adaptors, so one of my co-workers could take one back to Europe. I ordered on the 6th, paid for 2 day shipping, and got an e-mail that afternoon telling me my order status was shipped. Well, three days passed, and my co-worker flew back home without an IAXy. This week I asked voip-supply for a FedEx tracking number, and the tracking information shows the package actually shipped on the 13th, not the 6th like they told me! Their order status calculations are 7 days too optimistic. I sent them an e-mail about the status discrepancy, but I haven't received a response. I would consider their communication methods inadequate, unless you don't care when your order will arrive. -- Shaw Terwilliger [EMAIL PROTECTED] SourceGear LLC signature.asc Description: Digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo Canceller usage
Richard Scobie wrote: Or have I misunderstood how Zap EC works? Yes. The channels on the board are completely independent of each other. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disposition Failed still happening
Yeah, that's basically what's happening here too... our scenario: incoming call from PSTN --ZAP-- asteriskA --SIP-- asteriskB --SIP-- Phone asteriskA always shows answered for us, asteriskB sometimes shows answered, but it's usually failed... I've even gone through and started noop'ing the cause code to see what the server at least sees, and it always shows a code of 16... Aaron tracinet wrote: I should note that in the following scenario: incoming call from PSTN ---SIP--- asteriskA ---IAX2--- asteriskB ---SIP--- SIP Phone The call log does show disposition ANSWERED on asteriskA, but FAILED on asteriskB. On 12/15/05, *tracinet* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I actually opened a bug report on this earlier this month: http://bugs.digium.com/view.php?id=5918 I have tried a new SVN version from a few days ago and it still showed as FAILED for me in the following scenario: incoming call from PSTN ---SIP--- asterisk ---IAX2--- asterisk ---SIP--- SIP Phone At least now it appears that the billsec field is no longer showing zero, but the FAILED disposition is annoying. If I was a programmer I would happily jump in and see what could be done. Maybe in my free time I can squeeze in a lesson in C sometime ;) On 12/15/05, *Aaron Daniel* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Is anyone else still having disposition failed showing up in the cdr's on 1.2.1? I can't seem to figure out why asterisk would put that in the cdr's when the calls have in fact completed successfully 0.o Aaron Daniel ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: chan-capi avm b1 and capi.conf problems
Stefan Tichy wrote: On Thu, Dec 15, 2005 at 02:09:59PM +0100, Ricardo wrote: Dec 15 11:13:25 ERROR[4887]: chan_capi.c:4835 load_module: Unable to load config capi.conf, CAPI disabled You need /etc/isdn/capi.conf and /etc/asterisk/capi.conf. Also, check permissions on /dev/capi20 -- the process that runs Asterisk should have permission to that device, or else Asterisk won't start. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3202 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] about g729
On Dec 8, 2005, at 3:27 AM, Andrea Riela wrote: snipWith g711 all works like a charm, but for audio quality, and bandwidth utilization, I'm trying now to work with g729 between CME and ISP. What about Asterisk? this is a pass-thru example, or maybe I've to pay a g729 license? Yes, you need to buy the codec for $10(us) per channel if you want to be able to translate g729. I purchased the unsupported OSX version of the codec and it seems to work great and solved or improved many quality issues I was seeing. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Script to detect corrupted faxes from SpanDSP
#!/bin/bash # #Name: emailfax # Author: Colin Anderson [EMAIL PROTECTED] #Desc: Script to email faxes from SpanDSP and detect if fax is corrupt or incompatible with SpanDSP # These three variables must be passed to the script for it to work FAXFILE=$1 EMAILADDRESS=$2 CALLERID=$3 # First we convert the fax to a PDF wheter it's good, bad or whatever /bin/nice -n 19 tiff2ps -2eaz -w 8.5 -h 11 $FAXFILE | ps2pdf - $FAXFILE.pdf # Then we stat the filesize of the generated PDF. A corrupt PDF usually comes through as 422 bytes PDFSIZE=`stat -c%s $FAXFILE.pdf 2 /dev/null` # If-then to email the fax if it's OK, or email the recipient to let them know that the fax was bad, # and we will add it to our exception list (manually) so it will go to a real fax machine in the future # If the filesize of the PDF is greater than 422 bytes send it otherwise uh-oh. if [ $PDFSIZE -gt 422 ]; then mime-construct --to $EMAILADDRESS --subject Fax from $CALLERID --attachment $CALLERID.pdf --type application/pdf --file $FAXFILE.pdf rm $FAXFILE rm $FAXFILE.pdf else # I use mime-consruct because I'm lazy but a piped mail command should work just as well. mime-construct --to $EMAILADDRESS --subject Fax from $CALLERID failed to receive properly - this fax number will be added to the exception list mime-construct --to [EMAIL PROTECTED] --subject Fax from $CALLERID failed - fix dat shit rm $FAXFILE rm $FAXFILE.pdf fi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disposition Failed still happening
I have found that the only way asteriskA shows ANSWERED is if the call gets sent to an IVR on asteriskB and the caller hangs up before being connected to a SIP device. - PedroOn 12/15/05, Aaron Daniel [EMAIL PROTECTED] wrote: Yeah, that's basically what's happening here too... our scenario:incoming call from PSTN --ZAP-- asteriskA --SIP-- asteriskB --SIP-- PhoneasteriskA always shows answered for us, asteriskB sometimes shows answered, but it's usually failed... I've even gone through and startednoop'ing the cause code to see what the server at least sees, and italways shows a code of 16...Aarontracinet wrote: I should note that in the following scenario: incoming call from PSTN ---SIP--- asteriskA ---IAX2--- asteriskB ---SIP--- SIP Phone The call log does show disposition ANSWERED on asteriskA, but FAILED on asteriskB. On 12/15/05, *tracinet* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I actually opened a bug report on this earlier this month: http://bugs.digium.com/view.php?id=5918 I have tried a new SVN version from a few days ago and it still showed as FAILED for me in the following scenario: incoming call from PSTN ---SIP--- asterisk ---IAX2--- asterisk ---SIP--- SIP Phone At least now it appears that the billsec field is no longer showing zero, but the FAILED disposition is annoying. If I was a programmer I would happily jump in and see what could be done.Maybe in my free time I can squeeze in a lesson in C sometime ;) On 12/15/05, *Aaron Daniel* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Is anyone else still having disposition failed showing up in the cdr's on 1.2.1?I can't seem to figure out why asterisk would put that in the cdr's when the calls have in fact completed successfully 0.o Aaron Daniel ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
