[Asterisk-Users] Re: Call Screening
Hello-- I've done some coding for call screening in Asterisk. It's not in Asterisk yet, mainly because we're waiting for prompts from Allyson so it sounds like the rest of the system. But patches, prototype sound files, etc, are all filed at: http://bugs.digium.com/bug_view_page.php?bug_id=752 And I'd love to have your feedback. murf > Hi all. > > Is there a way to use asterisk for call screening? > > Meaning, a call comes in, asterisk answers with voicemail after I > don't > pickup, and the voicemail prompt + the caller's message a played via > the > sound card on asterisk. If I wan't to pick up, I do so by picking up > the > phone and dialing something. > Is it doable? > > Shoval Tomer, > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free World Dialup and Asterisk
Gonzalo,
Have you tried IAX, I see yo are behind NAT, and my experiences with IAX
behind NAT are much less painful :-)
I've FWD via IAX, receiveing calls (in fact, last time was a nearby
person in Portugal :-) that tested it).
One last thing, you mention dialup client, I guess she is not in dialup,
correct? From what I recall, FWD would do only G.711, would not exactly
work in dialup (maybe ISDN with 2 b-channels ?)
PS: I don't see the dialplan for the "inbound" calls, where a call from
FWD would land in your * ?
Gonzalo Gasca Meza wrote:
Hi forum,
I have been fighting days and days configuring FWD and asterisk with NO success
I have the following scenario.
My sister in Spain with FWD dialup client
My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone.
Spain LAN
FWD dialup account -> Internet <-- 3COM router/switch --- Asterisk -- 7960
I have done some research in google with no success.
http://www.m-networks.net/home/asterisk/ast-fwd.htm
http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
When I connect my FWD client in the LAN i can dial FWD numbers
ANY IDEAS OR CONF FILES WORKING WILL BE APPRECIATED
THANKS!
server*CLI> sip show registry
Host Username Refresh State
69.90.155.70:5060 431044 160 Registered
69.90.155.70:5060 421058 160 Registered
SIP.conf
register => 421058:[EMAIL PROTECTED]/103 ;Register Free World Dialup
register => 431044:[EMAIL PROTECTED]/103
[fwd1]
type=friend
username=431044
secret=password
fromuser=431044
fromdomain=fwd.pulver.com
host=fwd.pulver.com
insecure=very
canrenvite=no
nat = yes
dtmfmode=inband
[fwd2]
type=friend
secret=password
username=421058
fromuser=421058
fromdomain=fwd.pulver.com
host=fwd.pulver.com
dtmfmode=inband
nat=yes
canreinvite=no
extensions.conf
FWDUSERID1=421058
FWD1USERNAME=Gonzalo Gasca
FWDUSERID2=431044
FWD2USERNAME=Gonzalo Gasca
FWDPREFIX=*
[fwd1-out]
exten => _8.,1,SetCallerID(${FWDUSERID2})
exten => _8.,2,SetCIDName(${FWD2USERNAME})
exten => _8.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],70)
exten => _8.,4,Macro(fastbusy)
exten => _8.,5,Hangup
[fwd2-out]
exten => _7.,1,SetCallerID(${FWDUSERID1})
exten => _7.,2,SetCIDName(${FWD1USERNAME})
exten => _7.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],70)
exten => _7.,4,Macro(fastbusy)
exten => _7.,5,Hangup
My IP phone include those fwd1-fwd2-out
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Re: [Asterisk-Users] Phone choices....opinion request Polycom vs Cisco
> I am struggling with hardware choices to get started with. My options are > narrowed down to SIP phones - Polycom IP500, IP600 and Cisco 7940G. > > of importance is: > > - functionality / integration with asterisk > - headset functionality and use > - voice quality > - build quality > > Is there much of a difference between Polycom and Cisco? Scanning the group > it looks like there may be slightly more issues with Polycom but I don't > know how they stack up on the integration with Asterisk and future > flexability. > > Any recommendations appreciated. Each of the phones noted above are high quality business-class phones. You should be able to find user reviews on the wiki. The Polycom phones are a little bit more difficult to configure initially, but its a learning curve problem not a hardware/software problem. Polycom tends to use xml formated config files that are ftp'ed from your server after each phone reboot, so having a convenient way to view and edit those xml files will make your life easier. The Cisco sip phones use text files that are tftp'ed after each reboot to do the same thing, and those text files can easily be viewed/edited with any text editor. Far fewer options compared to Polycom, but the phones interoperate with asterisk just fine. The Polycom config files give you many more parameters to change/control compared to the Cisco. However, those added parameters is what makes the learning curve a little longer. The IP600 is probably the best choice from the above list since it tends to have everything that you could possibly want in a single phone including power over ethernet. The only down-side that I'm aware of is that you "must" obtain polycom's firmware from a reseller as polycom does not make those available to you on their web site. Not a big deal, but you need to be aware of it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting Siemens HiCom PBX with Asterisk through E1
Hi I've bought the Wildcard TE110 some days ago but I'm unable to get it to work with Siemens HiCom 300. I've tried this so far: 1. I've used standard cat5 cable cut off on one edge and twisted wires 1 to 4 and 2 to 3 which is according to cisco a short circuit for E1 lines. Wildcard takes a few seconds and sets the link to green (OK). 2. I've tried to connect our running E1 line from the telco with wildcard. The modem (modulates HDSL to G703 120 Ohms). I've used a 1:1 cable that did not work. I even tried to connect the copper wires by hand which resulted that the modem gave me a green power light but Wildcard stayed on a waving red light. 3. I have plugged out our running PBX and connected it to Wildcard which resulted in a green light for one second and then the state from zttool switched to yellow (and Wildcard to constant red light). The protocol used by the modem is hdb3 and /etc/zaptel.conf is adjusted according to this. Can anyone clarify the different protocol layers and when fails what? When occurs the green light? Must protocol layer2 be established or even higher or is it just a layer1 link light. Please help me out. Jens ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free World Dialup and Asterisk
He's trying to use sip, not iax. It would appear he's got both a fwd registration issue and an incoming fwd context issue. They don't appear to be in sync (probably an understanding of context issue actually). > Yes, of course you can do that. I have this very setup working for the > office, with * aggregating voip and isdn incoming calls and forwarding > them to my laptop wherever I am. > just follow the instructions on the FWD website, and run "iax2 debug" from > the console to see what's happening in anything goes wrong. > l. > > > In data Sat, 18 Dec 2004 20:33:01 -0800 (PST), Gonzalo Gasca Meza > <[EMAIL PROTECTED]> ha scritto: > > > Hi forum, > > I have been fighting days and days configuring FWD and asterisk with NO > > success > > I have the following scenario. > > My sister in Spain with FWD dialup client > > My question is if she can dial my FWD dialup number, which is registered > > in Asterisk and the call being forwarded to ring my IP Phone. > > > > Spain > > > > LAN > > FWD dialup account -> Internet <-- 3COM router/switch --- > > Asterisk -- 7960 > > -- > Creato con M2, il rivoluzionario client e-mail di Opera: > http://www.opera.com/m2/ > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] audio levels via sip
> I see from reading the mailing list theres a way to set audio levels on the > zap channels but I'm wondering if there's a way to set audio levels on > either sip or iax channels. I'm using some BT-100's and people are saying > the audio levels are a little low and I would like to bring them up a bit. The sip and iax channels do not have a rxgain/txgain option; only the zap channels have it. As someone else mentioned, some sip phones have gain adjustments. Not sure about the BT-100's as I don't use that phone. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free World Dialup and Asterisk
Inline...
> Hi forum,
> I have been fighting days and days configuring FWD and asterisk with NO
> success
> I have the following scenario.
>
> My sister in Spain with FWD dialup client
> My question is if she can dial my FWD dialup number, which is registered
> in Asterisk and the call being forwarded to ring my IP Phone.
>
> Spain
> LAN
> FWD dialup account -> Internet <-- 3COM router/switch --- Asterisk --
> 7960
>
> I have done some research in google with no success.
> http://www.m-networks.net/home/asterisk/ast-fwd.htm
> http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
>
>
> When I connect my FWD client in the LAN i can dial FWD numbers
> ANY IDEAS OR CONF FILES WORKING WILL BE APPRECIATED
> THANKS!
>
>
>
>
>
> server*CLI> sip show registry
> Host Username Refresh State
> 69.90.155.70:5060 431044 160 Registered
> 69.90.155.70:5060 421058 160 Registered
>
> SIP.conf
> register => 421058:[EMAIL PROTECTED]/103 ;Register Free World Dialup
> register => 431044:[EMAIL PROTECTED]/103
Change the above register statements to include you FWD number. Like:
register => 431044:[EMAIL PROTECTED]/431044
That register statement is telling FWD what extension to ring at _your_
location when FWD attempts to complete a call to your location.
> [fwd1]
> type=friend
> username=431044
> secret=password
> fromuser=431044
> fromdomain=fwd.pulver.com
> host=fwd.pulver.com
> insecure=very
> canrenvite=no
> nat = yes
> dtmfmode=inband
>
> [fwd2]
> type=friend
> secret=password
> username=421058
> fromuser=421058
> fromdomain=fwd.pulver.com
> host=fwd.pulver.com
> dtmfmode=inband
> nat=yes
> canreinvite=no
> extensions.conf
> FWDUSERID1=421058
> FWD1USERNAME=Gonzalo Gasca
> FWDUSERID2=431044
> FWD2USERNAME=Gonzalo Gasca
> FWDPREFIX=*
> [fwd1-out]
> exten => _8.,1,SetCallerID(${FWDUSERID2})
> exten => _8.,2,SetCIDName(${FWD2USERNAME})
> exten => _8.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],70)
> exten => _8.,4,Macro(fastbusy)
> exten => _8.,5,Hangup
>
> [fwd2-out]
> exten => _7.,1,SetCallerID(${FWDUSERID1})
> exten => _7.,2,SetCIDName(${FWD1USERNAME})
> exten => _7.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],70)
> exten => _7.,4,Macro(fastbusy)
> exten => _7.,5,Hangup
> My IP phone include those fwd1-fwd2-out
After making the register statement change noted above, add something
to your extensions.conf file to handle the incoming FWD call, like:
exten => 431044,1,Dial(SIP/103)
Or, another approach is something like this:
register => 431044:[EMAIL PROTECTED]
without the /431044 at the end, then in extensions.conf use:
exten => s,1,Dial(SIP/103)
so that all incoming calls from FWD match the "s".
The use the "sip debug" to watch calls and debug what is actually
happening.
Somewhere on the FWD web site is a url that will help diagnose the
problem. By clicking on their provided url, FWD will attempt to complete
a test call to your FWD number. It eliminates the need to have a
second party calling you for test purposes.
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[Asterisk-Users] Make asterisk launch script after completing call.
OK. I now have call recording working for both incoming and outgoing
calls.
Now I want to make those wavs into mp3. I could launch a script from
cron that checks for new wavs and converts them. But that wouldn't be
so elegant.
Launching it from * on hangup would be nicer. How is it done?
[outgoing]
exten => _0.,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten => _0.,2,Monitor(wav,${CALLFILENAME},m)
exten => _0.,3,Dial(SIP/rix/${EXTEN}|20|t)
exten => _0.,4,Congestion
exten => _0.,104,Congestion
[sip-in]
exten => 1000,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten => 1000,2,Monitor(wav,${CALLFILENAME},m)
exten => 1000,3,Dial(SIP/alex,20)
exten => 1000,4,Voicemail(u1000)
--
Alex Polite
http://polite.se
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Re: [Asterisk-Users] voicemailmain hotkey
On Sun, Dec 19, 2004 at 12:21:28AM -0600, Matthew Boehm wrote: > I'm having a similar problem. Do you have "operator=yes" in your > voicemail.conf under [general]? Argh, thats it, solved! Thanks a lot :) ...cut -- Tho/\/\as ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call billing
how do you integrate Gnugk and Asterisk billing? Are you using Asterisk's H323 channel?Voip Business <[EMAIL PROTECTED]> wrote: I integrate Gnugk and a gnugk billing system working like a charm.regardsHAOn Sat, 18 Dec 2004 01:48:56 -0800, Inam <[EMAIL PROTECTED]>wrote:> HI Alll> > this is my first post on users list> > can any body let me know how can one integrate his/her billing applications> to Asterisk Softswitch> > Thanks in advance> > INAMULLAH KHOSA> > ___> Asterisk-Users mailing list> [EMAIL PROTECTED]> http://lists.digium.com/mailman/listinfo/asterisk-users> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-users>___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream CallerID
> Is it possible to send the incoming PSTN caller ID to a Grandstream Budge > Tone-100 SIP phone? I've configured the extensions.conf file and the log is As Eric notes, the BT100 phones won't show letters. If a call comes in without CID, asterisk sends a string like "Asterisk call" which the BT will try to display as some giberish so I have setcallerid to "000" when this happens. I'd recommend using setcallerid(1234567890) (or any number) to test the phone, which should display that. If callerid does come in from PSTN, it should just make through as he said. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
> http://www.voip-info.org/wiki-RTP+Silence+Suppression > > http://lists.digium.com/pipermail/asterisk-users/2003-August/018670.html > > > So I am confused. The first says that VAD is supported in RTP. Ok, I know > that. The second is kinda ambiguous and seems to imply that * > doesnt support VAD. I think it does now as there is a VAD=yes option in > SIP.conf. > > Either way since IAX doesnt use RTP both statements are probably not > relevant. Does * support VAD with IAX? If so can it be turned > on and off in IAX?? Does anyone know definitively?? I really like to turn > it off and just send packet continuously. Should I file a bug > (feature request)?? Looking at the current /usr/src/asterisk/configs/sip.conf.sample, VAD=yes does not exist. Since those sample files tend to be the formal documentation for valid asterisk parameters, it should be safe to say its not supported. Same for iax.conf.sample; doesn't exist there either. The comment made by John Todd in the August 2003 posting was simply suggesting to the original poster (as that time) that he should enter a feature request into the asterisk bug tracker "if" he felt strongly that VAD was needed. The description of VAD in the voip-info reference is simply someone documenting what the sip rfc states about VAD. It does not imply or even suggest that asterisk supports VAD. Asterisk does not support VAD today (nor does it support every option documented in the sip rfc). The iax data flow betwen two boxes is not the same as sip-rtp data flows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phone choices....opinion request Polycom vs Cisco
Hi I am struggling with hardware choices to get started with. My options are narrowed down to SIP phones - Polycom IP500, IP600 and Cisco 7940G. of importance is: - functionality / integration with asterisk - headset functionality and use - voice quality - build quality Is there much of a difference between Polycom and Cisco? Scanning the group it looks like there may be slightly more issues with Polycom but I don't know how they stack up on the integration with Asterisk and future flexability. Any recommendations appreciated. Thanks Walt _ Don't just search. Find. Check out the new MSN Search! http://search.msn.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
> > I have * running on Mandrake 10.1 and I to had similar problems in the > > begging but as soon as I had ztdummy configured correctly everything > > seemed to just fall into place and work with IAX and *, not that I have > > got a perfect dialplan as that confuse's me but hey thats another subject. > > The problems you had and were resolved with ztdummy, were they primarily > IAX related ? > > Since, after all, the main channels relying on special timers are > Meetme, IAX and (maybe) MusicOnHold according to > http://www.voip-info.org/wiki-Asterisk+timer > > Just want to be sure, since I still believe my mere demo playback > issue likely has a different reason ... I'm 95% sure iax is not dependent on the ztdummy type timers. Maybe the OP could give us a little more detail on the specific data flow that he's having an issue with. I interpreted his call problem as: sipdev1 -> ? -> teliax.com -> iax -> OP-asterisk -> sipdev2 He indicated sipdev1 was running VAD, and the call was completed via teliax.com to his asterisk with crackly audio. If this "is" the case, the issue is VAD between sipdev1 and the "?" box shown in the data flow. Since there isn't a consistent flow of rtp data packets between sipdev1 and "?" because of VAD, what gets sent to teliax.com is already choppy audio. There is nothing the OP is going to be able to fix between teliax.com and sipdev2 to correct for a problem that is located elsewhere. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call waiting/ 3 way calling
HI; I have an Asterisk with 10 "SIP" ip-phones, our pbx features are now: Voicemail and Call Transfer. How can I serve both "Call Waiting / 3 way calling" for our SIP Phones.?/ Appreciate Any Help Mohammad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Per extension/user CDR?
It seems that all my CDR is dumping into the Master.csv file. There is a way to create per user/extension CDR but I have looked endlessly in the Wiki, docs, README.CDR, mailing list archives etc.. I can't seem to find a way to do this.. Any help would be appreciated. Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call screening
Sorry I mean the voice mail -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer Sent: Sunday, December 19, 2004 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] call screening Sorry, I don't follow. Dialing *98 will achieve what? Up until the time I decide to take the call, I want to be able to hear the person leaving a message interactively, so when I decide to pick up the call he's still there. Like a regular answering machine > -Original Message- > From: hadi [mailto:[EMAIL PROTECTED] > Sent: Sunday, December 19, 2004 12:13 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] call screening > > Yes > U can do it with asterisk and by dialing *98 on your Ip Phone you can > listen > to your message > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer > Sent: Sunday, December 19, 2004 1:40 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] call screening > > Hi all. > > Is there a way to use asterisk for call screening? > > Meaning, a call comes in, asterisk answers with voicemail after I don't > pickup, and the voicemail prompt + the caller's message a played via the > sound card on asterisk. If I wan't to pick up, I do so by picking up the > phone and dialing something. > Is it doable? > > Shoval Tomer, > IT Manager, > SofTov Advanced Systems, Ltd. > Office: +972-3-9230686 ext. 179 > Fax: +972-3-9216642 > Mobile: +972-54-8000200 > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > This message has been scanned for viruses and > dangerous content by MailScanner, and is > believed to be clean. > MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN HFC cards
Hi, Currently I am using a ISDN BRI PCI FRITZ card (works), would I get any benefits switching to a HFC card? Or it would be a better choice to switch to a ISDN with a DSP processor? Currently I have echo on my CAPI channel when calling analog lines, if call a cell phone, ISDN or a PRI PBX it doesn't show up any echo. So this indicates a far-end echo, how can this be minimized? I turned on the Squelch on the capi and it works but during a conversation the sometimes I tend to get small click and it distracts a little bit, even tough it is still better than the echo. If switching to HFC works better can someone point out where to buy them (online)? Regards, Humberto Aicardi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call screening
Sorry, I don't follow. Dialing *98 will achieve what? Up until the time I decide to take the call, I want to be able to hear the person leaving a message interactively, so when I decide to pick up the call he's still there. Like a regular answering machine > -Original Message- > From: hadi [mailto:[EMAIL PROTECTED] > Sent: Sunday, December 19, 2004 12:13 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] call screening > > Yes > U can do it with asterisk and by dialing *98 on your Ip Phone you can > listen > to your message > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer > Sent: Sunday, December 19, 2004 1:40 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] call screening > > Hi all. > > Is there a way to use asterisk for call screening? > > Meaning, a call comes in, asterisk answers with voicemail after I don't > pickup, and the voicemail prompt + the caller's message a played via the > sound card on asterisk. If I wan't to pick up, I do so by picking up the > phone and dialing something. > Is it doable? > > Shoval Tomer, > IT Manager, > SofTov Advanced Systems, Ltd. > Office: +972-3-9230686 ext. 179 > Fax: +972-3-9216642 > Mobile: +972-54-8000200 > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > This message has been scanned for viruses and > dangerous content by MailScanner, and is > believed to be clean. > MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call screening
On Sunday 19 December 2004 10:13, hadi wrote: > Yes > U can do it with asterisk and by dialing *98 on your Ip Phone you can > listen to your message No, that's voicemail. ie: The caller leaves a message and hangs up, then you retreive the message later. The OP wanted to be able to hear the incoming call as it happens, and then decide whether or not to pick up the phone and talk to the caller (much like you can do with many answering machines). Antony. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer > Sent: Sunday, December 19, 2004 1:40 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] call screening > > Hi all. > > Is there a way to use asterisk for call screening? > > Meaning, a call comes in, asterisk answers with voicemail after I don't > pickup, and the voicemail prompt + the caller's message a played via the > sound card on asterisk. If I wan't to pick up, I do so by picking up the > phone and dialing something. > Is it doable? -- I own three Windows books, published by O'Reilly. They are "Windows Annoyances", "Office 97 Annoyances" and "Windows 98 Annoyances". That pretty much sums it up for me. Please reply to the list; please don't CC me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call screening
Yes U can do it with asterisk and by dialing *98 on your Ip Phone you can listen to your message -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer Sent: Sunday, December 19, 2004 1:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] call screening Hi all. Is there a way to use asterisk for call screening? Meaning, a call comes in, asterisk answers with voicemail after I don't pickup, and the voicemail prompt + the caller's message a played via the sound card on asterisk. If I wan't to pick up, I do so by picking up the phone and dialing something. Is it doable? Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call screening
Hi all. Is there a way to use asterisk for call screening? Meaning, a call comes in, asterisk answers with voicemail after I don't pickup, and the voicemail prompt + the caller's message a played via the sound card on asterisk. If I wan't to pick up, I do so by picking up the phone and dialing something. Is it doable? Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
Martin List-Petersen wrote: On Sun, 2004-12-19 at 02:11, David Uzzell wrote: Then the other thing if mem serves me you are running 2.6 kernel so why not run ztdummy? With the 2.6 kernel this does not require any specialist Hardware or anything! Sorry, but maybe you should have read his posts more thoroughly. ztdummy is not an option because of his chipset. He has usb-ohci. ztdummy requires usb-uhci. Umm yes it does on 2.4 kernel but on a 2.6 kernel it doesn't cause I am running it on a 2.6 kernel and I don't have that hardware. Quoted from http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer "On kernel version 2.6 it uses internal high-resolution kernel timer and do not require any additional hardware. " Now in the original post he says that he is using FC2 so I am not 100% sure if it is 2.6 or 2.4 but FC2 is only one step away from FC3 which does run a 2.6 kernel. I don't know on FC2 as I have never run it. And yes to answer the original poster it did solve my IAX problems. With the demo I would sugest that maybe the SMP kernel on a single CPU server could be a partial cause. I have seen strange things on Dual CPU servers running SMP kernels were 1 CPU has been removed. Hope that helps. David Slán leat, Martin List-Petersen Dublin, Eire (contact info on --> http://www.marlow.dk/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDMoE or IAX?
On Sun, 19 Dec 2004, Eric Bishop wrote: > Apart from the the coolness factor can anyone explain to me in what > situation one would use TDMoE rather than IAX for communication > betwwen 2 Asterisk servers? There are two advantages with TDMoE: * low latency (prevents far end echo from going from nice sidetone to irritating percevied echo) * supports full pri signalling (hangupcause, type of number etc) There are disadvantages as well compared to iax: * non routeable (local ethernet only) * channels have to be preconfigured * more? I guess the key factor is if you need the low almost-tdm latency. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free World Dialup and Asterisk
Yes, of course you can do that. I have this very setup working for the office, with * aggregating voip and isdn incoming calls and forwarding them to my laptop wherever I am. just follow the instructions on the FWD website, and run "iax2 debug" from the console to see what's happening in anything goes wrong. l. In data Sat, 18 Dec 2004 20:33:01 -0800 (PST), Gonzalo Gasca Meza <[EMAIL PROTECTED]> ha scritto: Hi forum, I have been fighting days and days configuring FWD and asterisk with NO success I have the following scenario. My sister in Spain with FWD dialup client My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone. Spain LAN FWD dialup account -> Internet <-- 3COM router/switch --- Asterisk -- 7960 -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
Bruno Hertz wrote: On Sun, 2004-12-19 at 13:11 +1100, David Uzzell wrote: I have * running on Mandrake 10.1 and I to had similar problems in the begging but as soon as I had ztdummy configured correctly everything seemed to just fall into place and work with IAX and *, not that I have got a perfect dialplan as that confuse's me but hey thats another subject. The problems you had and were resolved with ztdummy, were they primarily IAX related ? Since, after all, the main channels relying on special timers are Meetme, IAX and (maybe) MusicOnHold according to http://www.voip-info.org/wiki-Asterisk+timer Just want to be sure, since I still believe my mere demo playback issue likely has a different reason ... I'd like to chime in here as I have a similar problem. I have been toying with * on other (cheapo) hardware not so successfully (mainly due to the audio chipsets). I just purchased an ASUS AV8 (Socket 939 Athlon 64 3500+) system for my real world testing, it's a high end MB and overall it has 98% of the feature set for what I wanted to accomplish. Currently I'm running FreeBSD 5.3 under the amd64 port of the OS (fyi). I'm experiencing the exact same symptoms - choppy clicking of the demo voice. I'll start by saying that I have done a reasonable amount of research on *, MB chipsets, and FreeBSD, and I've spent considerable time getting the basic functionality to work. The "ports" version of * under FreeBSD needed some tweaking to work under amd64 vs i386, but I have a working version including h323 and oss that works with the demo stuff. From what I have read the issue with choppy sound under the demo voice seems to be due to a timing issue, one that can't be solved under FreeBSD with the zaprtp (linux) stuff, and I haven't seen anything as far as USB stuff that will handle this. I do not have a Digium card installed yet, but I will have a TDM400P in a couple of days. Will a Digium card with the current driver solve the problem ? (zaptel doesn't compile for FreeBSD 5.3 amd64, maybe for i386). Given that I have a working installation with the same symptoms as reported, I'm leaning towards us having the same problem. If this is a timing issue, it would be great to solve this in a systematic way (without external hardware). Thoughts? Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
On Sun, 2004-12-19 at 02:11, David Uzzell wrote: > Then the other thing if mem serves me you are running 2.6 kernel so why > not run ztdummy? With the 2.6 kernel this does not require any > specialist Hardware or anything! Sorry, but maybe you should have read his posts more thoroughly. ztdummy is not an option because of his chipset. He has usb-ohci. ztdummy requires usb-uhci. Slán leat, Martin List-Petersen Dublin, Eire (contact info on --> http://www.marlow.dk/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
