Re: [Asterisk-Users] Placing call files in/var/spool/asterisk/outgoing/ does not work

[Jimmy Smith]

make sure that the shell location is right...#!/bin/bashnot equal to #!//ust/local/bincommon mistake.. On 5/25/06, Maxim Vexler
 <[EMAIL PROTECTED]> wrote:On 5/24/06, Michael Collins <
[EMAIL PROTECTED]> wrote:> > you should mv the file (and in the same filesystem, so 'rename' is> used)> >>> You might want to chmod or even chown the file first as well.  I wrote a
> little script that does all of this before the .call file is mv'd into> the outgoing directory:>> cp /tmp/test3.call /tmp/test1.call> chmod 666 /tmp/test1.call> chgrp asterisk /tmp/test1.call
> chown asterisk /tmp/test1.call> mv /tmp/test1.call /var/spool/asterisk/outgoing/>> I've been doing a lot of test calls to work on other aspects of .call> files, so this script is quite handy for making the same call over and
> over again.>> HtH>> -MC> ___> --Bandwidth and Colocation provided by Easynews.com -->
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>Thank you both, Tzafrir and Michael.I ended up using the script fromhttp://www.voip-info.org/wiki-Asterisk+tips+callbackIt looks like this:
<<<#!/bin/bash if [ "x$1" != "x" ];then phone_number="$1"; else return 1; fi  #note use of '-' in '-EOF1' - Escapes tab at beginning of lines  CALLFILE=$(cat <<-EOF1
Channel: Zap/g0/$phone_numberMaxRetries: 2# Retry in 5 minRetryTime: 300WaitTime: 45Context: ext-localExtension: 211Priority: 1EOF1)   echo "$CALLFILE" >> "/var/spool/asterisk/outgoing/max.call"
   # echo "$CALLFILE" >> "/tmp/calltomake.txt">>>I call it by doing a simple ./makecall.sh SOMEPHONEThe only problem I'm having now is that I must run the script as user
asterisk, otherwise I get the same asterisk error as before.Is there any way to fix this ?Thank you.--Cheers,Maxim Vexler"Free as in Freedom" - Do u GNU ?___
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[Asterisk-Users] Way to disable codec in dialingplan

[Jimmy Smith]

in a dialing plan.. extensions.confcan we enable or force a codec on specified npa..EX: 514NNN,1,force(gsm)  514NNN.2. dial(sip/blah)???
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[Asterisk-Users] cisco 7960

[Jimmy Smith]

does one know how to program so i can have 2 lines on one sip account on that phone ?im runnign my own asteriskdo i need 2 local accounts ? one for each line ? that rebounds to same SIP forp VOIP provider ?

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Re: [Asterisk-Users] cdrtool

[Jimmy Smith]

Ok anyone have latest cdrtool running 4.1 i think..
ill pay for install
On 1/31/06, Jimmy Smith <[EMAIL PROTECTED]> wrote:
i understand.. anyone know how much is basic support from them ?


On 1/31/06, [EMAIL PROTECTED] <
[EMAIL PROTECTED]> wrote:
Hello,Call 
sip:[EMAIL PROTECTED]
Regardsharry--- Jimmy Smith <[EMAIL PROTECTED]> a écrit :> anyone having weird problems on latest cdrtool?
>>> #!/usr/bin/php4
> *Fatal error*: Class> webservice_ngnprocdrtool_ngnprocdrtool: Cannot> inherit> from undefined class soap_client in> */var/www/CDRTool/SOAP/client_lib.php*on line> *2>

> always get weird error like that>> *> > ___> --Bandwidth and Colocation provided by 
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>___Nouveau
: téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs
exceptionnels pour appeler la France et l'international.Téléchargez sur http://fr.messenger.yahoo.com
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Re: [Asterisk-Users] cdrtool

[Jimmy Smith]

i understand.. anyone know how much is basic support from them ?


On 1/31/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
Hello,Call sip:[EMAIL PROTECTED]
Regardsharry--- Jimmy Smith <[EMAIL PROTECTED]> a écrit :> anyone having weird problems on latest cdrtool?>>> #!/usr/bin/php4
> *Fatal error*: Class> webservice_ngnprocdrtool_ngnprocdrtool: Cannot> inherit> from undefined class soap_client in> */var/www/CDRTool/SOAP/client_lib.php*on line> *2>
> always get weird error like that>> *> > ___> --Bandwidth and Colocation provided by Easynews.com> --
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[Asterisk-Users] cdrtool

[Jimmy Smith]

anyone having weird problems on latest cdrtool?



#!/usr/bin/php4


Fatal error:  Class webservice_ngnprocdrtool_ngnprocdrtool:  Cannot inherit from undefined class soap_client in /var/www/CDRTool/SOAP/client_lib.php on line 2

always get weird error like that


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Re: [Asterisk-Users] making wakeup feature call phone number, not extension?

[Jimmy Smith]

SIP/localext
or ZAP/someothr
or providers SIP/1xxxnnn
On 1/18/06, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> wrote:
With most asterisk installs, there's no difference between an extensionand a phone number.  For example, from your internal context, a phonecould dial one number and get an internal destination, or dial another
number and get an external destination.  i.e. in my office I could tella theoretical wake-up system to call me at extension 112 -- my desk --or at extension 7471234 -- my home.  Why should it need to know thedifference?  It's still the same procedure: call, wait for answer, play
info recording... :)MojoMoises Silva wrote:> I dont know the wakeup feature. But what you want can be done with a> web interface generating ".call" files with the timestamp of the day,
> hour and time when you want to hear the reminder. Just read in> voip-info about the ".call" files and if you have doubts we will be> glad to help you.>> Regards>> On 1/16/06, Roger Hanson <
[EMAIL PROTECTED]> wrote:>>>How would one go about setting up the wakeup feature of Asterisk to NOT>>call an extension, but to call a phone number?>>
>>My setup works great for wakeup on local extensions, but I'd like to set>>it up to call external phone numbers automatically and play a specific>>sound file (to remind people of upcoming hair stylist appointments).
I suppose either there'd have to be a web interface to use for this>>(entering a time for the reminder - and a phone number to call) or>>change the voice prompt to ask for a phone number to use, if not the
>>extension called from.I'm sure it's doable - but I am now knowledgeable enough.  I searched>>and didn't find any instructions on the web for something like this.>>
>>Thanks for any help...>>___>>--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list
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>--Mojo <[EMAIL PROTECTED]>Office Manger, Horan & Company, LLC(907) 747- x112___
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Re: [Asterisk-Users] ASterisk and home lines.. DGM-TDM01B or x100 ?

[Jimmy Smith]

i only see the spa1001 on voip-supply

http://www.voipsupply.com/product_info.php?manufacturers_id=14&products_id=320

can you confirm this is what i want ?

how would it work ? the phone jack on spa to the house patch and the rj45  into the asterisk hub
?

 
On 12/23/05, Ben Higley <[EMAIL PROTECTED]> wrote:
I do this..But you need the FXS module, not FXO.I personally use a Sipura 1000 connected to my house wiring for my phonesin the house.> but i could connect it to the home patch box and use any phoen in house
> right ? On 12/23/05, Ben Higley <[EMAIL PROTECTED]> wrote: the FXO is to connect to PSTN.. You need an FXS device (like a sipura 1000, 2000, or even 3000), or an
>> iaxy. etc. ./Ben > I got asterisk at home and not ma Bell, so i intend to use the>> internal>> > house wiring to use and connect a patch cable to my asterisk to the
>> house.>> >>> > this way i can pick up any extension in hous and will pop on zap>> device..>> >>> > now>> >>> > i got a x101p
>> >>> > Dec 19 09:26:07  kernel: wcfxo: DAA mode is 'FCC'>> > Dec 19 09:26:07  kernel: Found a Wildcard FXO: Wildcard X101P>> > Dec 19 09:26:07  kernel: ZapTel device loaded.
>> >>> >>> > can that do it ? or that needs an actualy bell signal ?>> > can i switch some weird asterisk config ?>> > ___
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Re: [Asterisk-Users] ASterisk and home lines.. DGM-TDM01B or x100 ?

[Jimmy Smith]

but i could connect it to the home patch box and use any phoen in house right ?

On 12/23/05, Ben Higley <[EMAIL PROTECTED]> wrote:
the FXO is to connect to PSTN..You need an FXS device (like a sipura 1000, 2000, or even 3000), or aniaxy. etc../Ben> I got asterisk at home and not ma Bell, so i intend to use the internal
> house wiring to use and connect a patch cable to my asterisk to the house.>> this way i can pick up any extension in hous and will pop on zap device..>> now>> i got a x101p
>> Dec 19 09:26:07  kernel: wcfxo: DAA mode is 'FCC'> Dec 19 09:26:07  kernel: Found a Wildcard FXO: Wildcard X101P> Dec 19 09:26:07  kernel: ZapTel device loaded.>>> can that do it ? or that needs an actualy bell signal ?
> can i switch some weird asterisk config ?> ___> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list
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[Asterisk-Users] ASterisk and home lines.. DGM-TDM01B or x100 ?

[Jimmy Smith]

I got asterisk at home and not ma Bell, so i intend to use the internal
house wiring to use and connect a patch cable to my asterisk to the
house.

this way i can pick up any extension in hous and will pop on zap device..

now

i got a x101p

Dec 19 09:26:07  kernel: wcfxo: DAA mode is 'FCC'
Dec 19 09:26:07  kernel: Found a Wildcard FXO: Wildcard X101P
Dec 19 09:26:07  kernel: ZapTel device loaded.


can that do it ? or that needs an actualy bell signal ? 
can i switch some weird asterisk config ?


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Re: [Asterisk-Users] sill looking for a provider

[Jimmy Smith]

ahaahah now thats something to be worry about.. that prolly coz they
dont want to pay taxes and your invoices serve you as refunds/credits
for irs..

BUT it is required by law to give irs that crap.. 

so i guess they pushing in offshore an will disappear someday with all the service and cash.

When something looks suspect it usually is. Ever seen smoke without any  fire ?
ps'' especialy the payment part" USTOMERS MAY NOT DISCLOSE USE OF OR
PAYMENTS TO VOIPJET ON PERSONAL, CORPORATE, LEGAL, ACCOUNTING AND

lol



On 11/7/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
OOPPS!  Looks like someone just broke voipjet's tosgw at adcomcorp.com gw at adcomcorp.com wrote onSat Nov 5 11:36:46 CST 2005
 I tend to agree with you, my experience with Teliax has been decent,and getting better.  If only I could get to them at under 20ms though,
right now my latency is about 75ms whereas voipjet comes through at19ms.Greg--
https://www.voipjet.com/tos.phpNON-DISCLOSURE:
ALL CUSTOMERS USING VOIPJET'S SERVICE ARE SPECIFICALLY PROHIBITED FROM
DISCLOSING TO OTHERS THAT THEY USE VOIPJET'S SERVICE, THIS INCLUDES BUT
IS NOT LIMITED TO, END USERS. CUSTOMERS MAY NOT DISCLOSE USE OF OR
PAYMENTS TO VOIPJET ON PERSONAL, CORPORATE, LEGAL, ACCOUNTING AND OTHER
DOCUMENTS AND COMMUNICATIONS UNLESS SPECIFICALLY REQUIRED TO DO SO BY
LAWHas anyone else read these TOS'es???  Some are pretty funny.Thomas HerlihyScaletta Moloney ArmoringChicago, IL USA708.924.0099Skype VoIP @ HerlsOne
Free World Dialup 647717[EMAIL PROTECTED]www.scaletta.com___--Bandwidth and Colocation sponsored by 
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Re: [Asterisk-Users] Re: Teliax billing question

[Jimmy Smith]

FYI 60/1 measn first 60 seoncd billed then each 1 /60th of a minute

so 1minute 25 second call is billed as 1 minute 25
  45 second as one minute
wher the first number is minimum seconds

so 6/6 is first 6 seconds no matter what then every block of 6 seconds..

6/1 well you get the point..

its like cell carriers..

where some bill per minute others like canada fido per second..

On 12/19/05, trixter aka Bret McDanel <[EMAIL PROTECTED]> wrote:
On Mon, 2005-12-19 at 10:13 -0800, Wolfgang S. Rupprecht wrote:> >> from: http://www.trac.org/news/2005/tracnotes-vol-3-22.html> >
> > The scam isn't new, and its certainly not limited to home 800 numbers.> > The same basic principles were used by some of the 900 number folks> > a few years ago as well.>> My fear wasn't that someone would stuff phony charges on my bill (like
> charges for 900 calls that were never made).  I was more afraid of the> case where someone in bad faith war-dials the 800 number so they can> collect the 60-cent (???) per call payphone charge.  Will VOIP
> providers let your dispute this charge because the calls were made in> bad-faith or is this simply a grin-and-bear it type situation?>That could be covered under 18 USC 1343 (wire fraud).  afaik there has
not been a single case that was prosecuted, and for the payphoneoperator (providing they meet the compensation requirements of the FCCrules - 13.65 comes to mind but I havent owned a payphone business since
1998 so I may not remember correctly) to make up some wild story abouthow it was a kid or something (which doesnt negate the payphoneoperators claim to compensation).  An elligible payphone must beavailable for the general public to get access to it.
All payments are typically made through clearinghouses as opposed toinidvidual carriers processing the billing.  This makes fraud trackingslightly easier since all the calls are there.  They have kept averages
of total calls by a payphone to compensatable numbers, carrier averages(ie mci, sprint, at&t, etc) and stuff that way.If someone were to use an auto dialer to call a tollfree they violate atleast 47 CFR 
64.1200 and I think a criminal statute too (I dont rememberwhere in the USC it is anymore, but there is one for that).According to the FCC rules back in 1997-98 on this matter even if fraudis suspected you must pay the payphone operator.  They also talk about
civil damages being sought, but that doesnt preclude criminal charges,only gives you easy rights to sue, which of course costs money and theburden of proof is then upon you.> I understand that within the PSTN there is a 2-bit value associated
> with the class of phone that the call is placed from (normal,> payphone, prison-phone).  If voip/pstn gateways started passing this> on it might make it easier for folks to guard against payphone scams
> by configuring their asterisk to only answer the 800 calls made from> normal residential phones.Any reasonable provider should be able to block those calls, however ina blocking situation its all or nothing.  If you have ani you can look
for the same number calling over and over and reject it that way.  Youshould have ani with a tollfree.The additional info is commonly not sent and afaik there is no'standard' way to send that.  SIP IM might work (that is how verisign
sends SS7 info in their SIP-7 product so doing something in this caseshouldnt be *too* hard but the provider has to agree to it).--Trixter http://www.0xdecafbad.com
 Bret McDanelUK +44 870 340 4605   Germany +49 801 777 555 3402US +1 360 207 0479 or +1 516 687 5200FreeWorldDialup: 635378http://www.sacaug.org/ Sacramento Asterisk Users Group
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Re: [Asterisk-Users] anybody getting "No authority found" with teliaxnow?

[Jimmy Smith]

maybe your account was disbaled due to non payment ?

also check if you are sending auth in the features area..

On 12/22/05, Jonathan k. Creasy <[EMAIL PROTECTED]> wrote:
This is an authentication problem. Check the username, password, numberand context being sent across to see if they are correct.Post your iax debug info for the call if you can.-Jonathan> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-> 
[EMAIL PROTECTED]] On Behalf Of Thomas Miller> Sent: Thursday, December 22, 2005 8:58 AM> To: asterisk-users@lists.digium.com> Subject: [Asterisk-Users] anybody getting "No authority found" with
> teliaxnow?>> Everything was working great until last night. All> calls since last night are getting "No Authority> Found" message. I am using IAX2>> Is anybody else having this problem?
>> Thx,> Tom>> __> Do You Yahoo!?> Tired of spam?  Yahoo! Mail has the best spam protection around> 
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Re: [Asterisk-Users] Teliax IAX problems -- Asterisk doesn't see answer

[Jimmy Smith]

hey 1.2 b2 hs bugs.
On 10/18/05, Adam Moffett <[EMAIL PROTECTED]> wrote:
- Original Message -> *From:* Rob Fugina [EMAIL PROTECTED]>> *To:* asterisk-users@lists.digium.com
> asterisk-users@lists.digium.com>> *Sent:* Monday, October 17, 2005 5:14 PM> *Subject:* [Asterisk-Users] Teliax IAX problems -- Asterisk
> doesn't see answer>> Not to point the finger at Teliax, but I'm having some unique> problems with their service that are as yet unexplained.>> Incoming calls are fine.
>> Outgoing calls don't work, though they did at one time.  As of> today, I'm running the latest code from CVS.>> -- Called teliax/1314321> -- Call accepted by 
208.139.204.245 > (format ulaw)> -- Format for call is ulaw> -- IAX2/teliax-18 is making progress passing it to IAX2/iay1-17
> -- Hungup 'IAX2/teliax-18'>> I can see from a tethereal network dump that Teliax is sending an> ANSWER, but Asterisk never notices.  Teliax starts sending audio.> The callee gets audio.  The caller gets nothing.
>> I can make the same call through Nufone, and it works fine.> Asterisk sees the ANSWER.>> But the network trace differs significantly between Nufone and> Teliax.  With Nufone, I see RINGING, 'stop sounds', then ANSWER.
> With Teliax, I don't see the RINGING frame or 'stop sounds' frame> at all.>> I don't know enough about the IAX protocol to know where the real> problem here is.  Can anyone help?  What other info can I provide
> that would help diagnose?>> Thanks,> Rob>I think I'm having a very similar issue.   Does setting jitterbuffer=offin iax.conf make the problem go away?___
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Re: [Asterisk-Users] teliax audio issues - response

[Jimmy Smith]

yes i got my mainstream * with teliax

no problem..

keep it ,ulaw,alaw, no jitter and ,open your ports, all good for 9 months !

On 10/19/05, Rich Adamson <[EMAIL PROTECTED]> wrote:
For those on the list using iax with teliax.com, the intermitant one-wayaudio problem that I reported to them received the following response:"We currently use our own version of 
1.2 with our own patches on our boxesand the iax code is updated. You cannot use jitterbuffers with g729 or gsmas this causes audio issues. So yes turning jitterbuffers will fix IAXissues."So much for standards; rfc, defacto or otherwise.
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Re: [Asterisk-Users] Looking por a provider to work with asterisk

[Jimmy Smith]

i use teliax for primary and bell for failover.

no neeed yet for failover ;)


they got theyre shit together...On 11/3/05, Jason Brashear <[EMAIL PROTECTED]> wrote:














Thank you Gleim I will look into that.

-Jason

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of 
Gleim, Jason
Sent: Friday, November 04, 2005
11:53 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Looking por a provider to work with asterisk



 

Jason,

 

Back in August there was a post of a
sip.conf and extensions.conf that would setup Asterisk to work with Vonage. I
haven't tried it yet but the user that posted reported success. Search
the archive for "Asterisk and Vonage" and you should be able to
find it or e-mail me off-list and I'll send you a copy.

 

J

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of 
Jason Brashear
Sent: Thursday, November 03, 2005
11:03 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Looking
por a provider to work with asterisk



 

I know about broadvoice.com


But are they the only solution?

I want to have two lines with Asterisk.

This is just a home install.

Believe it or not I have been using Vonage for about 2 ½
years and now I want to get rid of them to

Use and learn Asterisk.

Any help would be appreciated.

-Jason







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Re: [Asterisk-Users] Caller ID How does it get setup?

[Jimmy Smith]

oh and one tech from them support said they will be handling all cname soon .. like ETA 1 week


can't wait to get my FBI name out...
On 11/5/05, Rich Adamson <[EMAIL PROTECTED]> wrote:
> OK I am exhausted.> I can't seem to figure out how to send a caller ID along with a> Outbound call. < snip >>> Anyway, Outbound Caller ID Hos is this done?> I now use VoicePulse as my provider.
I'm not a VoicePulse user, so not sure if they even support it.Here's how I do it with teliax.com; Calls directed to Teliax.comexten => _1NX,1,Set(CallerIDnum=4024325395|a)
exten => _1NX,2,Set(CallerIDname=NPI|a)exten => _1NX,3,Dial(IAX2/teliaxout/${EXTEN})Some of the itsp's will accept the above and pass it out their PRIlinks. Other itsp's don't do support it at all. It depends a lot on
who they are interfacing with from call completion perspective. Inother words, for US PRI links, some carriers refuse to allow theircustomers to set the CallerID Num while others allow it.None of them typically support setting the CallerID Name however.
CallerID Name is usually done via the central office serving thecustomer that you called. They do a libd database lookup just priorto completing the call, and in most cases that database is configurable
by the telco's.The folks at teliax.com provide two mechanisms for their voip customersto handle CallerID. The method above is one of those, and the secondis provided through fields that I complete on their web site. I obviously
use the method shown above.One other comment, generally you can't set the CallerID Num to any oldarbitrary number that you might want. It has to be a real and usablenumber.___
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Re: [Asterisk-Users] all circiuts busy now. resolution?????

[Jimmy Smith]

bet its a difference in register timer..

On 11/5/05, John Fraser <[EMAIL PROTECTED]> wrote:
Hi all, I am using a TE405P with an E1 from the telco.  I am getting the allcircuits busy now message about 5% of the time on outbound calls from sipsoft phones.  Has there ever been a resolution to this problem?
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Re: [Asterisk-Users] SER+ASTERISK

[Jimmy Smith]

my bad you are.. lol didnt realize..

On 11/4/05, Jimmy Smith <[EMAIL PROTECTED]> wrote:
you could wait infinitely or try users list..On 11/4/05, harry gaillac <
[EMAIL PROTECTED]> wrote:
Hello Walter,The ser an asterisk run in the same box.What do you mean "redirect host + port :)"
Sip agents send sip requests to ser (outbound proxy)and this one to asterisk !sip agents are both registered on ser and asterisk.Please to explain me how asterisk redirect therequests.
Regards
Harry--- Walter Willis <[EMAIL PROTECTED]> a écrit :> the ser an asterisk run in the same box???
>> redirect host + port :)>>
>>> 2005/11/4, harry gaillac <[EMAIL PROTECTED]>:> >> > Hello,
> >> >> > I wish to setup this scheme:
> > ser-0.9.4> > asterisk-1.2-bêta> > polycom ip300 phones> >> >> > asterisk 5050--> > |firewall+nat|-192.168.> > ser 5060---> >
> > My ip phones use ser as outbound sip proxy and> > asterisk as sip registrar server.> > Ser Forward REGISTER requests to asterisk however> when> > a phone try to send an invite message then
> asterisk> > send icmp to private ip (host=dynamic in sip.conf)> >> > Is it possible to solve this problem ?> >> > Regards> > Harry> >> >
> >> >> >> >> >> >> >>___> > Appel audio GRATUIT partout dans le monde avec le
> nouveau Yahoo! Messenger> > Téléchargez cette version sur> http://fr.messenger.yahoo.com
> > ___
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Re: [Asterisk-Users] SER+ASTERISK

[Jimmy Smith]

you could wait infinitely or try users list..On 11/4/05, harry gaillac <[EMAIL PROTECTED]> wrote:
Hello Walter,The ser an asterisk run in the same box.What do you mean "redirect host + port :)"
Sip agents send sip requests to ser (outbound proxy)and this one to asterisk !sip agents are both registered on ser and asterisk.Please to explain me how asterisk redirect therequests.Regards
Harry--- Walter Willis <[EMAIL PROTECTED]> a écrit :> the ser an asterisk run in the same box???>> redirect host + port :)>>
>>> 2005/11/4, harry gaillac <[EMAIL PROTECTED]>:> >> > Hello,> >> >> > I wish to setup this scheme:
> > ser-0.9.4> > asterisk-1.2-bêta> > polycom ip300 phones> >> >> > asterisk 5050--> > |firewall+nat|-192.168.> > ser 5060---> >
> > My ip phones use ser as outbound sip proxy and> > asterisk as sip registrar server.> > Ser Forward REGISTER requests to asterisk however> when> > a phone try to send an invite message then
> asterisk> > send icmp to private ip (host=dynamic in sip.conf)> >> > Is it possible to solve this problem ?> >> > Regards> > Harry> >> >
> >> >> >> >> >> >> >>___> > Appel audio GRATUIT partout dans le monde avec le
> nouveau Yahoo! Messenger> > Téléchargez cette version sur> http://fr.messenger.yahoo.com> > ___
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Re: [Asterisk-Users] Moments of silence - take2

[Jimmy Smith]

seems every 10 sec something is happeneing on your network...

make sure your router is rebooted often if you have QOS on it has they tend to get behind on queues..

or UDP crc checksum failing in router.. that happened to me
on a linksys

your ping is ok 60 is good

i would also test my lan quality .. or wan.. 

some providers cut connections every xx seconds to deter peer sharing 

THE KEY HERE is you said 2 providers.. meaning i higlhy doubt its them.. 

1 ok 
2 no way..

its on your side.. solution

#1 try another router.
#2 try to do a line quality test see if its regular interval something is hapepning..

check your mta also..

On 11/4/05, Adam Moffett <[EMAIL PROTECTED]> wrote:
I'm sorry, that previous message might have made more sense if it hadall the information that I had intended to send.We are having moments of silence in the middle of phone calls.Generally it's not more than a few seconds, but it's still a nuisance.
Our IAX providers (we have 2) become unreachable for periods of 5-15seconds roughly 3 times an hour.  It happens to both providers, but notat the same time.  Below you'll find a log excerpt (cat messages | grep
teliax) with regards to that.I was wondering if the two issues are related.  Either way, does any onehave any experience with regards to silence in the middle of phonecalls?  What possible causes should I be looking at.
Nov  2 17:44:36 NOTICE[16535] chan_iax2.c: Peer 'teliax' is nowUNREACHABLE! Time: 68Nov  2 17:44:46 NOTICE[16535] chan_iax2.c: Peer 'teliax' is nowREACHABLE! Time: 69Nov  2 18:42:56 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now
UNREACHABLE! Time: 68Nov  2 18:43:06 NOTICE[16535] chan_iax2.c: Peer 'teliax' is nowREACHABLE! Time: 67Nov  2 20:03:17 NOTICE[16535] chan_iax2.c: Peer 'teliax' is nowUNREACHABLE! Time: 68Nov  2 20:03:27 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now
REACHABLE! Time: 67Nov  2 23:26:46 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now TOOLAGGED (2076 ms)!Nov  2 23:26:56 NOTICE[16535] chan_iax2.c: Peer 'teliax' is nowREACHABLE! Time: 68Nov  2 23:44:01 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now
UNREACHABLE! Time: 68Nov  2 23:44:11 NOTICE[16535] chan_iax2.c: Peer 'teliax' is nowREACHABLE! Time: 68Nov  2 23:58:16 NOTICE[16535] chan_iax2.c: Peer 'teliax' is nowUNREACHABLE! Time: 70Nov  2 23:58:26 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now
REACHABLE! Time: 73Nov  3 02:58:48 NOTICE[16535] chan_iax2.c: Peer 'teliax' is nowUNREACHABLE! Time: 67Nov  3 02:58:58 NOTICE[16535] chan_iax2.c: Peer 'teliax' is nowREACHABLE! Time: 69Nov  3 06:24:19 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now
UNREACHABLE! Time: 69Nov  3 06:24:29 NOTICE[16535] chan_iax2.c: Peer 'teliax' is nowREACHABLE! Time: 68Nov  3 06:29:31 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now TOOLAGGED (2086 ms)!Nov  3 06:29:42 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now
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[Asterisk-Users] asterisk callerid problems

[Jimmy Smith]

running 1.2 on both servers

from A to B to a 7960

the 7960 receives callerid as 

"NAME
usernameofsipuser"


i tried setting callerid etc before doing the dial

A to B is via iax B to 7960 is via sip of course

on B right before i dial 7960 i noop calleridnum and name and both populated ok

is this a but between 1.2 and cisco soft ?


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[Asterisk-Users] vm notif

[Jimmy Smith]

 hey all.. got a nice one.. got a cisco phone connected to
asterisk A .. withc connects to ASTERISK B    ... my VM
is on B..  is there a way to relay VM notif to cisco ? 


thanks
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Re: [Asterisk-Users] Lots of console; attach and grep?

[Jimmy Smith]

hehe yes exactly..

you could tail -f that file .. or grep
as in 

tail -f /var/log/asterisk/verbose |grep -10 -v 'somestring'

that would give you 10 lines around it.. or before it i dont remmeber
off the bat..


On 8/24/05, Eric Wieling aka ManxPower <[EMAIL PROTECTED]> wrote:
> Matthew Boehm wrote:
> > We have recently started routing about 3 PRI's worth of traffic thru our
> > asterisk box.
> >
> > The text on the console now flys by so damn fast, I can't really see
> > what the heck is going on. Even with verbosity 0 and debug 0 it is still
> > so fast.
> >
> > Is there some way I can attach to the console in a way that will allow
> > me to grep or otherwise filter the text so I can focus on something in
> > particular?
> 
> You mean like the info in /var/log/asterisk which is configured via
> /etc/asterisk/logger.conf ?
> 
> 
> --
> Eric Wieling * BTEL Consulting * 504-210-3699 x2120
> 
> r: Generate a ringing tone for the calling party, passing no audio from
> the called channel(s) until one answers. Use with care and don't insert
> this by default into all your dial statements as you are killing call
> progress information for the user. Really, you almost certainly do not
> want to use this. Asterisk will generate ring tones automatically where
> it is appropriate to do so. "r" makes it go the next step and
> additionally generate ring tones where it is probably not appropriate to
> do so.
> 
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Re: [Asterisk-Users] Cisco 7960 / SIP & tftp configs

[Jimmy Smith]

line1_shortname: "Home"
line1_displayname:"Home"


On 8/24/05, Asterisk User Group <[EMAIL PROTECTED]> wrote:
> I have three questions about my 7960 phone that I can't discern from the
> docs/wiki.
> 
> 1st - If I change the SIPxx.cnf file to change registrations it sets
> up new lines as expected. If I delete a line it doesn't get removed when
> I reboot the phone. I have to go to the phone, unlock it, and reset the
> SIP parameters. How do I make it "forget" what it has programmed and
> listen only to the download?
> 
> 2nd - Has anyone figured out how to get the Message button to launch a
> dial to VoicemailMain?
> 
> 3rd - How do I display on the LCD an alias to the registered line?
> line1_name: 2000
> line1_authname: "2000"
> line1_password: **
> 
> The doc seems to suggest that line1_name is what it registers with and
> line1_authname is what it uses "if challenged during the
> authentication". This doesn't make any sense to me. I am looking for the
> line to be "2000" but the display to say "Home" or "Business", etc.
> 
> Thanks, dbc.
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Re: [Asterisk-Users] Lots of console; attach and grep?

[Jimmy Smith]

or verbose..

makse sure its enabled in logger.conf



On 8/24/05, Jimmy Smith <[EMAIL PROTECTED]> wrote:
> how about /var/log/asterisk/mssages
> 
> On 8/24/05, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> > We have recently started routing about 3 PRI's worth of traffic thru our
> > asterisk box.
> >
> > The text on the console now flys by so damn fast, I can't really see
> > what the heck is going on. Even with verbosity 0 and debug 0 it is still
> > so fast.
> >
> > Is there some way I can attach to the console in a way that will allow
> > me to grep or otherwise filter the text so I can focus on something in
> > particular?
> >
> > Thanks,
> > -Matthew
> >
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Re: [Asterisk-Users] Lots of console; attach and grep?

[Jimmy Smith]

how about /var/log/asterisk/mssages

On 8/24/05, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> We have recently started routing about 3 PRI's worth of traffic thru our
> asterisk box.
> 
> The text on the console now flys by so damn fast, I can't really see
> what the heck is going on. Even with verbosity 0 and debug 0 it is still
> so fast.
> 
> Is there some way I can attach to the console in a way that will allow
> me to grep or otherwise filter the text so I can focus on something in
> particular?
> 
> Thanks,
> -Matthew
> 
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Re: [Asterisk-Users] RealTime ignoringswitch=>Realtime/[EMAIL PROTECTED] me_ext

[Jimmy Smith]

mae sure you dont have skip-networking in my.cf for mysql

also make sure you can connect to ip , name and localhost form cmd
line first using these credentias



On 8/24/05, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> John Novack wrote:
> 
> >> Both modules should be able to use either hostname, IP, or local
> >> socket. They both use the same mysql API code to connect to the server.
> >>
> > Should perhaps, but they DON'T!
> 
> I will test this but again, they use the same mysql_real_connect() to
> connect to the server.
> 
> > Why  isn't there a good example of that on the Wiki or in the docs?
> > If you understand how to do that, then TEACH others!
> 
> It says it quite plainly in all the sample configs that if you are
> running a database on the same machine that you can use sockets.
> 
> -Matthew
> 
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[Asterisk-Users] Sql Realtime

[Jimmy Smith]

Anyone got 

Realtime with mysql database using myodbc-iodbc

and getting errors like SqL FETCH ERROR  on load ( 50 q /s ) on db ?


my db has 500 threads ready
this happens sometimes.. not always


im not sure if its asterisk -> odbc connector
or
odbc connector to mysql

but i assume its the first case...

.. options i found would be to use mysql  from the asterisk distro..
but are the memory leaks fixed ?

any opinions ?? ideas ?
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Re: [Asterisk-Users] problems with eyebeam - video phone

[Jimmy Smith]

Aug 17 08:37:06 WARNING[14731]: Don't know any of 0x8 formats

is from what you pasted btw..

Don't know any of 0x8 formats 

is
524288 (1 << 19)  (0x8)  videoh263   (H.263 Video)

meaning it downst understand it or find it



On 8/17/05, Jimmy Smith <[EMAIL PROTECTED]> wrote:
> quickly this looks like a incompatible codec.. or unrecognized..
> 
> show codecs on CLI>
> 
> show show
>  262144 (1 << 18)  (0x4)  videoh261   (H.261 Video)
> 524288 (1 << 19)  (0x8)  videoh263   (H.263 Video)
>1048576 (1 << 20) (0x10)  video   h263p   (H.263+ Video)
> does it ?
> 
> On 8/17/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> > Thank you for your answer.
> > I didn't register on the domain of the Eyebeam software, actually I don't
> > understand how to do that!
> > I bouught 5 eyebeam activation keys and I am trying with the first 2 of
> > them
> >
> > On the Eyebeam side (both eyebeam), I only enabled the "Basic H.263" codec,
> > no other.
> >
> > If, on the asterisk side in sip.conf, I put the gsm codec BEFORE h263, the
> > two video phone speak without any problem (but without any video)
> > If, on the asterisk side in sip.conf, I put the gsm codec AFTER h263, the
> > first video phone call the second, the second answer and immediately
> > the call ends.
> >
> > If Ilook at /var/log/asterisk/full, I see:
> > 
> > Aug 17 08:37:06 VERBOSE[14731]: -- AGI Script dialparties.agi
> > completed, returning 0
> > Aug 17 08:37:06 VERBOSE[14731]: -- Executing Dial("SIP/551-eac0",
> > "SIP/552|25|tr") in new stack
> > Aug 17 08:37:06 DEBUG[14731]: SIMPLE DIAL (NO URL)
> > Aug 17 08:37:06 DEBUG[14731]: Setting NAT on RTP to 0
> > Aug 17 08:37:06 DEBUG[14731]: Setting NAT on VRTP to 0
> > Aug 17 08:37:06 WARNING[14731]: Don't know any of 0x8 formats
> > Aug 17 08:37:06 DEBUG[14731]: Outgoing Call for 552
> > Aug 17 08:37:06 DEBUG[14731]: Call from user '552' is 1 out of 0
> > Aug 17 08:37:06 VERBOSE[14731]: -- Called 552
> > Aug 17 08:37:06 DEBUG[13529]: (Provisional) Stopping retransmission (but
> > retaining packet) on '[EMAIL PROTECTED]'
> > Request 102: Found
> > Aug 17 08:37:06 VERBOSE[14731]: -- SIP/552-ff46 is ringing
> > Aug 17 08:37:10 DEBUG[13529]: Acked pending invite 102
> > Aug 17 08:37:10 DEBUG[13529]: Stopping retransmission on
> > '[EMAIL PROTECTED]' of Request 102: Found
> > Aug 17 08:37:10 DEBUG[13529]: build_route: Contact hop:
> > 
> > Aug 17 08:37:10 VERBOSE[14731]: -- SIP/552-ff46 answered SIP/551-eac0
> > Aug 17 08:37:10 WARNING[14731]: No path to translate from SIP/551-eac0(2)
> > to SIP/552-ff46(524288)
> > Aug 17 08:37:10 WARNING[14731]: Had to drop call because I couldn't make
> > SIP/551-eac0 compatible with SIP/552-ff46
> > Aug 17 08:37:10 DEBUG[14731]: update_user_counter(552) - decrement outUse
> > counter
> >
> >
> > It seems the problem documented in bug
> > http://bugs.digium.com/bug_view_page.php?bug_id=0003709
> > but actually it is not exactly the same.
> >
> > moreover: is there any way to put the patch described in
> > http://bugs.digium.com/bug_view_page.php?bug_id=0003709 (enable H263p in *)
> > in asterisk 1.0.9 and not asterisk CVS HEAD ?
> >
> > Any help will be greatly appreciated.
> >
> > Andrea
> >
> >
> >
> >
> > "Carlos Alperin"
> > <[EMAIL PROTECTED]
> > om.net>To
> > Sent by:  "'Asterisk Users Mailing List -
> > asterisk-users-bo Non-Commercial Discussion'"
> > [EMAIL PROTECTED] 
> > m.com  cc
> >
> >   Subject
> > 16/08/2005 20.48  RE: [Asterisk-Users] problems with
> >   eyebeam - video phone
> >
> > Please respond to
> >  Asterisk Users
> >  Mailing List -
> >  Non-Commercial
> >Discussion
> > <[EMAIL PROTECTED]
> > ists.digium.com>
> >
> >
> >
> >
> >
> >
> > Hi,
> >
> > I get Eyebeam working with an older version of Asterisk 1.0.2(I believe). I
&

Re: [Asterisk-Users] problems with eyebeam - video phone

[Jimmy Smith]

quickly this looks like a incompatible codec.. or unrecognized..

show codecs on CLI>

show show 
 262144 (1 << 18)  (0x4)  videoh261   (H.261 Video)
 524288 (1 << 19)  (0x8)  videoh263   (H.263 Video)
1048576 (1 << 20) (0x10)  video   h263p   (H.263+ Video)
does it ?

On 8/17/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Thank you for your answer.
> I didn't register on the domain of the Eyebeam software, actually I don't
> understand how to do that!
> I bouught 5 eyebeam activation keys and I am trying with the first 2 of
> them
> 
> On the Eyebeam side (both eyebeam), I only enabled the "Basic H.263" codec,
> no other.
> 
> If, on the asterisk side in sip.conf, I put the gsm codec BEFORE h263, the
> two video phone speak without any problem (but without any video)
> If, on the asterisk side in sip.conf, I put the gsm codec AFTER h263, the
> first video phone call the second, the second answer and immediately
> the call ends.
> 
> If Ilook at /var/log/asterisk/full, I see:
> 
> Aug 17 08:37:06 VERBOSE[14731]: -- AGI Script dialparties.agi
> completed, returning 0
> Aug 17 08:37:06 VERBOSE[14731]: -- Executing Dial("SIP/551-eac0",
> "SIP/552|25|tr") in new stack
> Aug 17 08:37:06 DEBUG[14731]: SIMPLE DIAL (NO URL)
> Aug 17 08:37:06 DEBUG[14731]: Setting NAT on RTP to 0
> Aug 17 08:37:06 DEBUG[14731]: Setting NAT on VRTP to 0
> Aug 17 08:37:06 WARNING[14731]: Don't know any of 0x8 formats
> Aug 17 08:37:06 DEBUG[14731]: Outgoing Call for 552
> Aug 17 08:37:06 DEBUG[14731]: Call from user '552' is 1 out of 0
> Aug 17 08:37:06 VERBOSE[14731]: -- Called 552
> Aug 17 08:37:06 DEBUG[13529]: (Provisional) Stopping retransmission (but
> retaining packet) on '[EMAIL PROTECTED]'
> Request 102: Found
> Aug 17 08:37:06 VERBOSE[14731]: -- SIP/552-ff46 is ringing
> Aug 17 08:37:10 DEBUG[13529]: Acked pending invite 102
> Aug 17 08:37:10 DEBUG[13529]: Stopping retransmission on
> '[EMAIL PROTECTED]' of Request 102: Found
> Aug 17 08:37:10 DEBUG[13529]: build_route: Contact hop:
> 
> Aug 17 08:37:10 VERBOSE[14731]: -- SIP/552-ff46 answered SIP/551-eac0
> Aug 17 08:37:10 WARNING[14731]: No path to translate from SIP/551-eac0(2)
> to SIP/552-ff46(524288)
> Aug 17 08:37:10 WARNING[14731]: Had to drop call because I couldn't make
> SIP/551-eac0 compatible with SIP/552-ff46
> Aug 17 08:37:10 DEBUG[14731]: update_user_counter(552) - decrement outUse
> counter
> 
> 
> It seems the problem documented in bug
> http://bugs.digium.com/bug_view_page.php?bug_id=0003709
> but actually it is not exactly the same.
> 
> moreover: is there any way to put the patch described in
> http://bugs.digium.com/bug_view_page.php?bug_id=0003709 (enable H263p in *)
> in asterisk 1.0.9 and not asterisk CVS HEAD ?
> 
> Any help will be greatly appreciated.
> 
> Andrea
> 
> 
> 
> 
> "Carlos Alperin"
> <[EMAIL PROTECTED]
> om.net>To
> Sent by:  "'Asterisk Users Mailing List -
> asterisk-users-bo Non-Commercial Discussion'"
> [EMAIL PROTECTED] 
> m.com  cc
> 
>   Subject
> 16/08/2005 20.48  RE: [Asterisk-Users] problems with
>   eyebeam - video phone
> 
> Please respond to
>  Asterisk Users
>  Mailing List -
>  Non-Commercial
>Discussion
> <[EMAIL PROTECTED]
> ists.digium.com>
> 
> 
> 
> 
> 
> 
> Hi,
> 
> I get Eyebeam working with an older version of Asterisk 1.0.2(I believe). I
> only use H.263 and SIP. (G.729)
> 
> Now, the more important question is if you register on the domain on the
> Eyebeam software. I found that this was the full secret about this.
> 
> Let me know your configuration on the Eyebeam side.
> 
> Regards,
> 
> Carlos Alperin
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
> Sent: Tuesday, August 16, 2005 11:28 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] problems with eyebeam - video phone
> 
> I am trying to connect two Xten eyeBeam Video Phone
> 
> No problems in voice connecting.
> 
> I tryed to modify my sip.conf
> 
> [general]
> language=it
> videosupport=yes
> ; enable Asterisk video support
> 
> port = 5060   ; Port to bind to (SIP is 5060)
> bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
> disallow=all
> allow=h263
> allow=gsm
> allow=ulaw
> allow=alaw
> ; H.263 is our video codec
> ; allow=h263p
> ; H.263p is the enhanced video codec
> context = from-sip-external ; Send unknown SIP callers to this context
> callerid = Unknown
> 
> #include sip_nat.conf
> #include sip_custom.conf
> #include sip_additional.conf
> 
> An

Re: [Asterisk-Users] realtime caching

[Jimmy Smith]

pruning breaks asterisk on high loads

at least on all 5 of our servers.

all using different versions and custom.



"
What you can do is use "sip prune realtime " to remove just the
single peer/user from memory. And you can force a reload of that peer
from realtime by using "sip show peer  load".

"

On 8/17/05, Damon Estep <[EMAIL PROTECTED]> wrote:
> > > It seems that some options are not re-read when caching is on, for
> > > example, changing the caller ID value in the sip table has no effect
> > > until a reload (or expiration), so at least in some cases
> rtcahcefriends
> > > makes realtime notsorealtime.
> >
> >   No. It is doing exactly what it says it will, "cacheing". If you
> > have
> > rtcachefriends turned on, when a peer/user registers the info is
> pulled
> > from DB and added to the internal (a la 'in memory') list that
> chan_sip
> > maintains. If you change something in DB after this occurs then your
> > changes won't take affect because chan_sip has no need to re-lookup
> your
> > phones info since the info is already present in memory.
> >
> >   What you can do is use "sip prune realtime " to remove
> just
> > the
> > single peer/user from memory. And you can force a reload of that peer
> > from realtime by using "sip show peer  load".
> >
> >   If you want pure realtime where chan_sip always pulls from db,
> then
> > turn caching off. Keep in mind that turning caching off will remove
> MWI
> > and NAT functionality.
> >
> > -Matthew
> >
> What would it take (you, $) to add functionality that is a cross between
> caching and not, that is it caches with a flag in the extension, so if
> the flag is present realtime will be queried even though the extension
> is in cache when a new call comes IN TO that extension.
> 
> Outgoing calls would not really need a re-query unless something about
> the provisioning of the phone changes, at which point it would
> re-register anyways, right?
> 
> The goal is caching for MWI and NAT but realtime for calling, so the
> database is checked on every inbound call in case the dialplan changed,
> and the cache updated accordingly.
> 
> Maybe a TTL flag, and when the TTL expires the cache entry stays, but is
> re-queried when a dialplan match is found. The admin could then tune the
> performance by setting different TTLs, maybe 15 minutes for lightly
> loaded systems, 4 hours for heavy loaded systems.
> 
> Dynamic updates take place in whatever timeframe is specified on the TTL
> or less.
> 
> Have I missed something, is this functionality already present?
> 
> Damon
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Re: [Asterisk-Users] URGENT: hardware spesifications needed

[Jimmy Smith]

coudnt agree more.. thats exact thing i was saying the other day..

please hold my di..k while i take a leak i don't want to wet my hands.

RTFM, google and test. || Pay

On 7/6/05, Brian West <[EMAIL PROTECTED]> wrote:
> Why not do your research instead of asking the list to do it for
> you  lazy ass!
> 
> /b
> ---
> Anakin: "You're either with me, or you're my enemy."
> Obi-Wan: "Only a Sith could be an absolutist."
> 
> On Jul 6, 2005, at 2:09 AM, Erdem HAKİ wrote:
> 
> > Hello;
> >
> >
> >
> > I need to set up Asterisk to serve and register for 1000 users(not
> > simultaneus). What kind of specifications do my server need.
> >
> >
> >
> > For example:
> >
> >
> >
> > Xenon processor
> >
> > 1 GB RAM
> >
> > 120 GB HDD  etc...
> >
> >
> >
> > Thanks for your help..
> >
> >
> >
> > Erdem HAKI – [EMAIL PROTECTED]
> >
> > ___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
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Re: [Asterisk-Users] converting windows .wav to .gsm

[Jimmy Smith]

i can suggest using wavepad.


its on the voipinfo site

On 7/6/05, mohammad <[EMAIL PROTECTED]> wrote:
>  
> HI ALL; 
>   
>   
> I have problem converting a windows .wav file to .gsm format by Sox. 
> Could anyone help. 
>   
>   
> Cheers, 
> Mohammad 
>   
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>
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[Asterisk-Users] cdrtool anyone ?

[Jimmy Smith]

does cdrtool handle 800 termination from different src ?

from the page they say
"Combined rating based on traffic, duration, application type and destination"



so not from src it seems..

anyone got that working ?

Example : billing depending on src number + destination #

JV
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[Asterisk-Users] cdrtool

[Jimmy Smith]

does cdrtool handle 800 termination from different src ?

from the page they say
"Combined rating based on traffic, duration, application type and destination"



so not from src it seems..

anyone got that working ?

Example : billing depending on src number + destination #

JV
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Re: [Asterisk-Users] Asterisk-CVS-HEAD locks up on 'reload' from CLI (sometimes)

[Jimmy Smith]

this happens to me too.

On 7/6/05, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Wed, Jul 06, 2005 at 11:48:27AM -0500, Brian West wrote:
> >
> > /b
> > ---
> > Anakin: "You're either with me, or you're my enemy."
> > Obi-Wan: "Only a Sith could be an absolutist."
> >
> > On Jul 5, 2005, at 12:02 PM, Chris Coulthurst wrote:
> >
> > >Lately when I issue a 'reload' from the CLI, I find that it will
> > >sometimes
> > >hang forever, completely locked up.  I can press enter and see the CLI
> > >prompt move, but no commands are taken.  "top" shows asterisk eating
> > >everything up:
> 
> What exactly is it doing?
> 
> attach to it with strace (strace -p) or with ltrace to get some clues.
> 
> [snip]
> 
> >
> > rm -rf /usr/include/asterisk
> >
> > do a fresh checkout and try again.
> 
> Mind giving some details, apart from a "reinstall" kind of
> advice? To allow prevension? Or for those who have some extra files in
> /usr/include/asterisk
> 
> Thanks
> 
> --
> Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
> http://tzafrir.org.il |   | a Mutt's
> [EMAIL PROTECTED] |   |  best
> ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] Simpletelecom dead?

[Jimmy Smith]

6  beyond-the-network.LosAngeles.savvis.net (208.173.57.30)  33.966 ms
 34.143 ms  33.841 ms
 7  * * *

hangs there...

savvis invoice paid ?

beyond-the-network a black hole ?


On 7/4/05, Gary Reuter <[EMAIL PROTECTED]> wrote:
> Hmmm
> Can't place calls...
> Can't access website...
> Neither of the 3 nameservers answer anything...
> Anyone heard/know something to explain all this?
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Re: [Asterisk-Users] Long delay via Teliax

[Jimmy Smith]

or use an iax client..
many are out there


http://voip-info.org/tiki-index.php?page=VOIP+Phones



On 7/4/05, Joseph <[EMAIL PROTECTED]> wrote:
> On Mon, 2005-07-04 at 17:48 -0400, Jimmy Smith wrote:
> > another example of what i was saying..
> >
> > connect directly if still does this its not you my friend..
> >
> > people should really stop using asterisk as first connect attempts to
> > test a service.
> >
> > use a direct client on provider .,
> >
> > Asterisk is complicated with many settings, unix flavors , hardware
> > and bandwith could affect.
> >
> > i got 4 dids on teliax as well as many if not all providers under an
> > alias for testing wholesale
> > ill need.
> >
> > never had a problem.
> > then again i always started to test a provider with xten then moved to
> > a more complicated platform.
> >
> > WHEN i have problems i recheck by isolating Asterisk out of the loop
> > untilll im sure its not on my side.
> >
> > MHO
> 
> It is hard to take the asterisk out of the loop as the 800-number is
> coming over IAX; so the only solution as Chris suggested it to  analyze
> the network path for latency.
> 
> --
> #Joseph
>
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Re: [Asterisk-Users] Proper way to start * and load modules on a RedHatbox

[Jimmy Smith]

how about /etc/rc.local

#a line that would work
path/to/screen -d -m path/to/asterisk -vvgfc






  -d -m   Start screen in "detached" mode. This creates a new session but
   doesn't  attach  to  it.  This  is  useful  for  system startup
   scripts.



On 7/4/05, Carlos Alperin <[EMAIL PROTECTED]> wrote:
> Did you check the log files looking for load errors?
> 
> Carlos Alperin
> Senior System Engineer
> Seneca Communications, LLC
> [EMAIL PROTECTED]
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende
> Sent: Monday, July 04, 2005 3:06 PM
> To: Asterisk Users List
> Subject: [Asterisk-Users] Proper way to start * and load modules on a
> RedHatbox
> 
> Hi list!
> 
> I have several boxes running asterisk as I want, no problems there but the
> one thing I haven't sorted out is how to properly start asterisk on boot
> time.
> 
> This is probably a n00b class question but how do I properly set this up
> (I didn't find any docs on this).
> 
> The zaptel script alone does not load the proper driver on boot time, I
> guess I need to do some thing with the alias stuff in modules.conf?
> 
> Also how can I make the startup scripts appear in ntsysv? Even when I copy
> the scripts to rc.d they do not show up in ntsysv
> 
> I tried loading the modules manually from rc.local but that doesn't work,
> even if I use delays. For some reason ztcfg doesn't work when run from
> rc.local and therefore asterisk fails to load. If I run ztcfg manually
> then ztcfg starts properly.
> 
> Thanks for any hints / tips!
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Re: [Asterisk-Users] Long delay via Teliax

[Jimmy Smith]

another example of what i was saying..

connect directly if still does this its not you my friend..

people should really stop using asterisk as first connect attempts to
test a service.

use a direct client on provider ., 

Asterisk is complicated with many settings, unix flavors , hardware
and bandwith could affect.

i got 4 dids on teliax as well as many if not all providers under an
alias for testing wholesale
ill need.

never had a problem.
then again i always started to test a provider with xten then moved to
a more complicated platform.

WHEN i have problems i recheck by isolating Asterisk out of the loop
untilll im sure its not on my side.

MHO



On 7/4/05, Joseph <[EMAIL PROTECTED]> wrote:
> I'm testing Teliax tall free number line and I'm experiencing long delay
> about 1sec. during conversation.
> When I call myself over FWD the response is normal no delay or cut
> messages.
> When I call my number over FWD the is a long delay, welcome message
> usually cuts off few first words and during conversation my voice
> arrives after about 1sec. delay.
> Since, the 800-number is only accessible from USA and I'm in Canada, the
> only way I can test it is by calling it over FWD.
> 
> I've tested codec: ulaw and gsm
> What might be causing such a problem?
> 
> --
> #Joseph
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[Asterisk-Users] VOIP Providers Problems

[Jimmy Smith]

you guys are so friggin funny..

all i see bout problems on most providers here are users who never
read a line of the handbook

i could prolly solve all these eyes closed with the asterisk handbook
on my side as a friend.


wake up..

i work for a hosting provider and we get lots of users assuming they
have it all right and us is the problem. and in fact 99% of the time
they are the ones who fucked theyre servers up and such.

not to mention calling support to blah blah they fishing trips using
everybodys time for nothing.

and might i add the tickets , emails and posts on boards wich not only
is non sense but stupid of theyr part.



#1
EX : sending to all emails there is of a company
.

now how the fuck would aol know if one never answered or did if you
send to sales, support, marketing, etc etc all at once.

#2 one problem at a time..

people send 44 emails at once asking for support on basic shit like
how do i reboot my box.


then dispatch a second one for how do i compile a program..

RTFM


#3 posting in boards -  obsiouvsely we won't respond immediately to
these non support requests so user will post in forums saying we are
lame support or any other non sense.


all this just makes me wonder what the hell is the point of even
trying to get good customer service when  10 complaints will overcome
1000 good reviews.

And this from BS users who don't even lift a finger to help themselves
but rather put the load on staff..


my july 4 th 2 cents on why i would prefer having 100 biz clients then
10,000 users
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