Re: [asterisk-users] DIDForSale July Special (No Activation on new DID Purchases)

Sincere Apologies-- Send the mail to wrong list, Meant to send to asterisk-biz list. -J On Wed, Jul 8, 2009 at 11:35 PM, Jai Rangi wrote: > *All, > To meet the target for the month, we are running a special promotion. > > $5 activation fee waived for all new DID purchases.* > > Buy DIDs from D

[asterisk-users] DIDForSale July Special (No Activation on new DID Purchases)

*All, To meet the target for the month, we are running a special promotion. $5 activation fee waived for all new DID purchases.* Buy DIDs from DIDForSale today and *your $5 activation fees will be WAIVED* for all the DIDs purchased before July 20 2009. There is no lim

Re: [asterisk-users] Anonymous Connection form IP to use specific Context

Hi All, I never saw a reply to this question. Is anyone able to assist? Regards David. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Klaverstyn Sent: Friday, 19 June 2009 2:28 PM To: 'Asterisk Users Mailing List - Non-Commerci

Re: [asterisk-users] Dial stops trying after ~30s regardless

Hi - yes, you are correct in that I am using AMI. I thought I could override inline in the dialplan. I will modify the AMI call. Thanks for the quick response - truly appreciated. john -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.dig

Re: [asterisk-users] Dial stops trying after ~30s regardless

On 9/7/09 2:06 PM, John Regal wrote: > Hi, > > My Dial() is set to the following, but always stops about 30 seconds > into the call even when I set it to try for 60 seconds. > > exten => dialnumber,1,Dial(${DialInfo},60) > > I am running on 1.6.1-r199820. > > Is there some other setting that is ove

Re: [asterisk-users] q: am i mixing somethign up?

On 9/7/09 1:39 PM, tom wrote: > hi, > > checking my freshly installed astersik-gui, i can see a menu entry > called "Users". clicking on that one gives me the pages labeled (on > orange) "User Extensions on PBX". if i do make an entry here, it ends up > in the user.conf. file. > > so i created a ne

Re: [asterisk-users] calculate data traffic

On 9/7/09 12:11 AM, jonas kellens wrote: > To calculate the monthly data traffic that is generated by VoIP-calls, > is it as simpel as > > 80kbps (G.711 SIP) x 6s (1000 minutes) = 480 kilobits / month = > 585.9375 MB traffic / month http://www.asteriskguru.com/tools/bandwidth_calculator.ph

[asterisk-users] Dial stops trying after ~30s regardless

Hi, My Dial() is set to the following, but always stops about 30 seconds into the call even when I set it to try for 60 seconds. exten => dialnumber,1,Dial(${DialInfo},60) I am running on 1.6.1-r199820. Is there some other setting that is overriding mine? Or an issue with this release? Th

[asterisk-users] q: am i mixing somethign up?

hi, checking my freshly installed astersik-gui, i can see a menu entry called "Users". clicking on that one gives me the pages labeled (on orange) "User Extensions on PBX". if i do make an entry here, it ends up in the user.conf. file. so i created a new entry in the sip.conf, reloaded asterisk >

[asterisk-users] q: sip registration fails...

[Jul 8 21:23:49] WARNING[4358]: chan_sip.c:10458 check_auth: username mismatch, have <6001>, digest has <1160> [Jul 8 21:23:49] NOTICE[4358]: chan_sip.c:18529 handle_request_register: Registration from '>' failed for '192.168.1.3' - Username/auth name mismatch sip.conf [6001] user=6001 type = f

Re: [asterisk-users] g.722 + loudness

Hose wrote: > What you say...Dave Fullerton (dfullertaster...@shorelinecontainer.com): > > >> Kevin P. Fleming wrote: >> >>> Hose wrote: >>> >>> I have a feeling that the issue is between transcoding of ulaw to g.722 and it's too loud during the transcoding - anyway to adju

Re: [asterisk-users] What is the best way to share extension state

It does which is why it was not included in a release code set. The patch could be changed to do an OR type compare for the bridge class. I have changed my implementation to use only user events for everything that I now need so I did not pursue this patch. -- Jim Dickenson mailto:dicken...@cfmc.c

Re: [asterisk-users] g.722 + loudness

What you say...Dave Fullerton (dfullertaster...@shorelinecontainer.com): > Kevin P. Fleming wrote: > > Hose wrote: > > > >> I have a feeling that the issue is between transcoding of ulaw to g.722 > >> and it's too loud during the transcoding - anyway to adjust the levels? > > > > There was a fla

Re: [asterisk-users] Queue autopause

Christian Gansberger escribió: > Hi all! > > I want to autopause my queue member when they are not answering within > 20 seconds, and the autopause > should affect all queues they are member of, not only the queue where > the call was not answered. > > Is there a way to do that? > > The members get

Re: [asterisk-users] Using a mobile phone via USB as an extension

Thank you for the info Does anyone know if the cdc-modem interface which is available on mobile phones can actually potentially be used to initiate, or register for receiving a voice call? If so, I suppose USB 3G dongles could even be used as a voip-air interface! Would be interesting to find

Re: [asterisk-users] Caller ID (name) - where does it come from?

Barry D. Hassler wrote: > Well, Teliax says they "have no access to the PSTN's database", but > I'm suggesting they check out TargusInfo as mentioned above. One of > their suggestions, is to contact the local ILEC to get the number > published in their white pages. Will that accomplish the same

Re: [asterisk-users] q: install asterisk + asteris-gui

That's a new one on me, but check out this link http://forums.digium.com/viewtopic.php?t=3689 &highlight=&sid=acbc25fd45bae1ecc42b0d7ca66fe88c As I read it, you want to be able to dial 1001 and

Re: [asterisk-users] q: install asterisk + asteris-gui

thx danny, (sorry, bad day today) one more question: "deviceandusers" i had this distinction with freepbx, though i dont know whether this is a freepbx-thing or an asterisk-setting... thx ___ -- Bandwidth and Colocation Provided by http://www.api-digita

[asterisk-users] Queue autopause

Hi all! I want to autopause my queue member when they are not answering within 20 seconds, and the autopause should affect all queues they are member of, not only the queue where the call was not answered. Is there a way to do that? The members gets dynamically added. I'm using asterisk 1.4.21.2

Re: [asterisk-users] q: install asterisk + asteris-gui

If you're just going to use Asterisk as an internal system, you just need a simple users.conf, sip.conf and about a 5 line dialplan. Sip.conf [general] srvlookup=yes ;allows DNS lookups of server names naxexpirey=180 defaultexpirey=160 context=default ; Default context for incoming calls

Re: [asterisk-users] q: install asterisk + asteris-gui

thx again, one last question: as i mentioned, i used freepbx before. now i facing only the section: - users > my goal right now is to use that asterisk instance just to have intenral extensions to talk to each other...whats the quickest setup here? i mean i dont need trunks, dialplans etc, right?

[asterisk-users] Fwd: q: install asterisk + asteris-gui: SOLVED

:8088/asterisk/static/config/index.html wes my missing link thx 2 all for ur help -- Forwarded message -- From: tom Date: Wed, Jul 8, 2009 at 4:19 PM Subject: Re: [asterisk-users] q: install asterisk + asteris-gui To: Asterisk Users Mailing List - Non-Commercial Discussio

Re: [asterisk-users] q: install asterisk + asteris-gui

You're confusing the manager interface with the gui interface. The gui interface would be 8088/asterisk/static/config/index.html _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tom Sent: Wednesday, July 08, 2009 3:19 PM To:

Re: [asterisk-users] q: install asterisk + asteris-gui

yeah thx i did that. now if i log in ( :8088/asterisk/static/ajamdemo.html) , i see the Asterisk™ AJAM Demo. but thats it: i tries the urls givin by : "http show status", but none of them gives me a real webinterface to administrate the whole asterisk etc i thought asterisk-gui gives me the

Re: [asterisk-users] Grandstream 2010 and blinky lights

Because DEVSTATE is for custom hints - and have you tried to set one every time a phone rings/is answered? This was thought about - but the logic in the dialplan would be a nightmare. Anyway, doing it the way I do it works for me (and others) as my dialplan contains nothing but 'include' and 'swi

Re: [asterisk-users] q: install asterisk + asteris-gui

In http.conf make bindaddr be the address of your asterisk server. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tom Sent: Wednesday, July 08, 2009 3:01 PM To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercia

Re: [asterisk-users] q: install asterisk + asteris-gui

stupid me, i had a ; in front of the [general] line. thx so far im logged inand now? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.

Re: [asterisk-users] q: install asterisk + asteris-gui

thx, but still struggeling: http://blabla:8088/asterisk/static/docs/index.html >> NO GO --- ; ; Asterisk Builtin mini-HTTP server ; ; ; Note about Asterisk documentation: ; If Asterisk was installed from a tarball, then the HTML documentation should ; be installed in the stati

Re: [asterisk-users] q: install asterisk + asteris-gui

Do you have the [general] section with enabled, webenabled, port and ipaddress? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tom Sent: Wednesday, July 08, 2009 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Disc

Re: [asterisk-users] q: install asterisk + asteris-gui

/etc/manager.conf: [admin] secret = test read = system,call,log,verbose,command,agent,user,config write = system,call,log,verbose,command,agent,user,config - doenst let me log in ;-( - i tried chown /static_http/config this is in my apache-logs: [Wed Jul 08 15:36:23 2009] [error] [client 66.134

Re: [asterisk-users] q: install asterisk + asteris-gui

On Wed, 2009-07-08 at 14:49 -0400, tom wrote: > - repointes apache /var/www/1234 >> /var/lib/asterisk/static_html The Asterisk GUI uses the web server built into Asterisk, so what you're attempting to do here isn't going to work. I suggest you follow the instructions at http://astbook.asteriskdoc

Re: [asterisk-users] q: install asterisk + asteris-gui

shame on me...yes i had several different installations of asterisk, just to try it out. but i deleted everything before i went on installing a different version or vendor. so, make samples did the trick! i now have the missing files. thx (i didnt do it before coz somehow samples + freepbx) screwd

Re: [asterisk-users] q: install asterisk + asteris-gui

Since /etc/asterisk is empty, you have either relocated your conf files or put them in a database. Assuming neither, just create manager.conf in /etc/asterisk with this setup [general] Enabled = yes Port = 5038 Webenabled=yes Bindaddr = 1.2.3.4 [loginname] Secret=secret And restart a

Re: [asterisk-users] q: which Browser-GUI do u guys use?

On Wed, 8 Jul 2009, tom wrote: > > None. I'm a command line weenie. ) GUIs don't let you annotate your changes -- who did what (or what they thought they were doing), when, and why. ) GUIs don't support any sort of "versioning." ) GUIs don't support any sort of configuration rollback. All of

Re: [asterisk-users] q: which Browser-GUI do u guys use?

- "tom" wrote: > > *MY* browser must be experiencing problems. I thought you posted a message but it appears blank. I'm a huge fan of elinks. It's cross platform and works great. --Tim ___ -- Bandwidth and Colocation Provided by http://ww

Re: [asterisk-users] q: install asterisk + asteris-gui

- "tom" wrote: > hi, i > @asterisk > - svn-ed asterisk from digium 1.6 > - make install > > >> its running and i can access the CLI > > @gui > then i > -svned asterisk-gui from digium > - installed > - repointes apache /var/www/1234 >> /var/lib/asterisk/static_html > >> now, i see

[asterisk-users] q: which Browser-GUI do u guys use?

___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] q: install asterisk + asteris-gui

hi, i @asterisk - svn-ed asterisk from digium 1.6 - make install >> its running and i can access the CLI @gui then i -svned asterisk-gui from digium - installed - repointes apache /var/www/1234 >> /var/lib/asterisk/static_html >> now, i see the login box, but i dont have any credentials. tutorial

Re: [asterisk-users] Automatic Gain Control

Danny Nicholas wrote: > If you are using a large number of DAHDI channels, you could designate a > chunk of them as "non-local" since you can control RXGAIN on each channel. > You would have to work out something with your TELCO since your'e a dead > duck control-wise once you answer the call. >

Re: [asterisk-users] MixMonitor/Queue and Tranfers

Un-topposting... > On Tue, Jul 7, 2009 at 7:08 PM, Miguel Molina > wrote: > >> Darrin Henshaw escribió: >> >> 2. The issue does seem to be limited to MixMonitor and the Queue >> application, as in testing I setup mixmonitor on my extension dialed it from >> outside the company(my cell

Re: [asterisk-users] Caller ID (name) - where does it come from?

Well, Teliax says they "have no access to the PSTN's database", but I'm suggesting they check out TargusInfo as mentioned above. One of their suggestions, is to contact the local ILEC to get the number published in their white pages. Will that accomplish the same thing (I doubt it). On Wed, Jul 8,

Re: [asterisk-users] Calling non-extension numbers issue

The two "logs" that I have been able to find are messages on the asterisk server in debug. Unfortunately, Nokia does not have any kind of logging (sucks). What I can see is that it is definitely a phone issue, just stuck on where to go from here. First, this if from asterisk in debug 1

Re: [asterisk-users] g.722 + loudness

Kevin P. Fleming wrote: > Hose wrote: > >> I have a feeling that the issue is between transcoding of ulaw to g.722 >> and it's too loud during the transcoding - anyway to adjust the levels? > > There was a flaw in Asterisk's G.722 transcoder module that was fixed > recently (on May 15, 2009), so

Re: [asterisk-users] Small site survivability

Audiocodes supports SRST on their mediapack analog gateways. This might be a viable option. I haven't used any Audiocodes devices before. Are people pleased with them? Deploy a lot of small asterisk based appliances... > > This way you can completely decentralise your setup and give each o

[asterisk-users] Grandstream GXP-1200 & G.722?

Can anyone here have experience using G.722 on the Grandstream GXP-1200? It's supposed to support the codec, but I wonder if the handset does it justice? The older BT-200 also supported the codec, but the handset was not good enough. You could only hear the improved call quality using a headset.

Re: [asterisk-users] false answer on zaptel

On Mon, 06 Jul 2009 10:31:18 -0500 Brent Davidson wrote: > Botond Botyanszki wrote: > > Hi, > > > > I have an x100p zaptel card with asterisk 1.4. I'm using the system for > > outgoing calls. > > My problem is that Answer() is falsely returning while the call is still > > ringing and was not rea

Re: [asterisk-users] Restarting of B-channel on span 1

add resetinterval=never in your zaptel.conf, or chan_dahdi.conf depending on what you are running. zaptel or dahdi. On Wed, Jul 8, 2009 at 10:35 AM, Aman Dhally wrote: > Hi All, > > Hope you all are fine and good, Today i have found that Mine all PRI > Channels are restating after every interval

Re: [asterisk-users] g.722 + loudness

Hose wrote: > I have a feeling that the issue is between transcoding of ulaw to g.722 > and it's too loud during the transcoding - anyway to adjust the levels? There was a flaw in Asterisk's G.722 transcoder module that was fixed recently (on May 15, 2009), so any release made after that date sho

[asterisk-users] Restarting of B-channel on span 1

Hi All, Hope you all are fine and good, Today i have found that Mine all PRI Channels are restating after every interval of one hour, and i have search and psot on fourms and everyone said that this is a normal behaviour. If this is a normal behaviour is there is any way to stop it { i s

Re: [asterisk-users] g.722 + loudness

Hose wrote: > Hi, > > > >

Re: [asterisk-users] Caller ID (name) - where does it come from?

CALLERID(name) is a TELCO specific field. In the long run, you will be best served using your own lookup of a database using CALLERID(num), since CID(name) is unreliable and in some cases costly. IMO, you would be well served with an app (AGI?) that recorded valid names into the database and let

Re: [asterisk-users] Play a recorded message when a fax is detected ?

You should initiate a second call or send a voicemail. You don't want to mess too much with what is working. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Tuesday, July 07, 2009 1:32 PM To: Asterisk Users Ma

Re: [asterisk-users] Automatic Gain Control

If you are using a large number of DAHDI channels, you could designate a chunk of them as "non-local" since you can control RXGAIN on each channel. You would have to work out something with your TELCO since your'e a dead duck control-wise once you answer the call. -Original Message- From:

Re: [asterisk-users] Call parking with ISDN

The sort of trunk does matter; I don't know about ISDN, but I get different behavior on DAHDI vs SIP, so that's one verification that you are dealing with a necessarily fixed set of values. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lis

Re: [asterisk-users] MixMonitor/Queue and Tranfers

Thanks for the reply. 1. The extensions in the Queues are setup as Agent members, defined in Agents.conf, then within the definition of the queue in queues.conf they are made members of the queue. 2. As for the recording my diaplan is as follows: [main-line] exten => s,1,NoOp() exten => s,n,NoOp

Re: [asterisk-users] SALE 70% OFF on Pfizer

Title: asterisk-users@lists.digium.com • Wed, 8 Jul 2009 03:34:15 +0100

[asterisk-users] calculate data traffic

To calculate the monthly data traffic that is generated by VoIP-calls, is it as simpel as 80kbps (G.711 SIP) x 6s (1000 minutes) = 480 kilobits / month = 585.9375 MB traffic / month ??? Jonas. ___ -- Bandwidth and Colocation Provided by http:/

Re: [asterisk-users] Caller ID (name) - where does it come from?

On 7/8/09, Steve Totaro wrote: > On Wed, Jul 8, 2009 at 2:14 AM, Olivier wrote: > > Hi, > > > > Reading this thread, is this correct to say CallerName is widely used in > the > > US ? > > > > Here in France, this service is optional but I don't think many companies > > are subscribing to i

[asterisk-users] asterisk + cisco as5400 t.38 fax sending.

Hello, I heard that since asterisk 1.6.0.6, now you can send faxes with t.38 through asterisk to a PST gateway that supports t.38 too. Is that true ? If so, what elements you need to make it work beside asterisk and the PSTN trunk ? Thanks all.- ___ --

Re: [asterisk-users] Caller ID (name) - where does it come from?

On Wed, Jul 8, 2009 at 2:14 AM, Olivier wrote: > Hi, > > Reading this thread, is this correct to say CallerName is widely used in the > US ? > > Here in France, this service is optional but I don't think many companies > are subscribing to it and I'm not aware of any non-Telco CNAM providers. > I w

Re: [asterisk-users] asterisk addon mysql - is mysql connection persistent

Thanks Miguel Molina :) I was bit curious about that as I am using few asterisk boxes connected to a mysql server. And that mysql server sometimes gets lots of connections from other sides ( other than asterisk boxes) . So if asterisk-mysql holds dedicated persistant connection , it means cdr are

Re: [asterisk-users] Asterisk and Skype

when using sisky you could integrate an ivr menu Alex Balashov schreef: This is not currently possible. Work in progress. -- Sent from mobile device On Jul 8, 2009, at 1:31 AM, DHAVAL INDRODIYA wrote: Hello All, can anybody tell me how can i integrate asterisk and skype users so th

Re: [asterisk-users] Asterisk & Jabber : WARNING: res_jabber.c [RESOLVED]

This is my jabber.conf : [general] debug=yes ;;Turn on debugging by default. ;autoprune=no ;;Auto remove users from buddy list. ;autoregister=yes ;;Auto register users from buddy list. [asterisk]

Re: [asterisk-users] Grandstream 2010 and blinky lights

On 8/7/09 8:52 PM, Andrew Thomas wrote: > That's exactly the way I do it as well :D > > > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian > Lyndon-Smith > Sent: 06 Jul

Re: [asterisk-users] Grandstream 2010 and blinky lights

That's exactly the way I do it as well :D -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: 06 July 2009 11:16 To: Asterisk Users Mailing List - Non-Com

Re: [asterisk-users] Asterisk and Skype

DHAVAL INDRODIYA wrote: > Hello All, > > can anybody tell me how can i integrate asterisk and skype users > > so that skype users can dial my asterisk number or dial internal > dialplan form skype > > regars > Dhaval > Chan_celiax can apparently interface with a copy of the skype client runni

Re: [asterisk-users] Automatic Gain Control

> 17. Automatic Gain Control (Brent Davidson) > Is there any possibility of DAHDI supporting Automatic gain control on > TDM ports? Have a look at asterisk-1.6.1 and module func_speex.so, which provides AGC function. This function can be applied to any channel. Documentation: http:

Re: [asterisk-users] documentation of DAHDI dial options

Jared Smith schrieb: > On Tue, 2009-07-07 at 15:42 +0200, Klaus Darilion wrote: >> I am searching for the description of the available dialstrin options >> for the DAHDI channel (and also other channel types). >> >> I am not looking for outdated voip-info links, but for the authoritative >> sou