Re: [asterisk-users] [ASTERISK-USER] Meetme Recording issue, recording is 2 times Faster then normal recording.
Please check WAV headers, what is the sample rate of the file? It should be 8kHz. Does the WAV sound normal when you decrease sample rate by hand? You can just upload one WAV for testing - I'll say what may be wrong with it. On Tue, May 18, 2010 at 9:52 AM, DHAVAL INDRODIYA wrote: > hello All, > > i have one issue with Asterisk Meetme Application > > i am recording through Meetme channels through option 'r' and format for > recording a file is 'wav' > > lines used is DAHDI version 2.1.0.4. and asterisk version 1.6.0.5. > > i have very strange problem of meetme_recording , > > once conference starts recording file having a recording is 2x faster than > normal recording . > > is there any setting to solve it out , my card type is TE410P used E1 lines > . > > please help me . any help appreciated. > > regards > Dhaval > > > > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ?
2010/5/18 Gopalakrishnan A.N > you can use SNOM VoIP phones > Have you tried them with 100 BLF ? For instance, Aastra phones are limited to 50 BLF (though you can have much more buttons). > > On Tue, May 18, 2010 at 11:58 AM, Olivier wrote: > >> Hi, >> >> Can you share successful experience with a SIP hardphone supporting 100 >> BLF ? >> Which phone would you suggest for that ? >> >> (In case that matters, each BLF is supposed to SUBSCRIBE to and reflect >> the state (Idle, Ringing, OnCall) of a local extension. >> >> Regards >> >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > Thank you with regards, > Gopalakrishnan A.N, > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] automon filename does not follow the docs.
Hi there, We used to record all the calls with the Monitor function. Now, I haveimplemented on-demand recording with automon instead... Everything is working fine apart from the generated filename, which as per all docs, should be auto-epoch-caller-calleebut in my case, it is auto-epoch-who_started_record-the_other_end. Thank you all in advance. Regards, Marta -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.30 & T38
Hello list,
I read on voip-info.org that Asterisk 1.4 support T38 passthrough.
So I guess this means that I can have a Grandstream HT503 with T38
support and an analogue faxmachine on the other side of my Asterisk and
a T38-account with a ITSP on the other side of my Asterisk machine, right ?!
The fax coming from the faxmachine passes the HT503 to my Asterisk and
my Asterisk sends the fax to the ITSP which places it on the PSTN.
What do I place in the dialplan ??
The only dialplan I find is :
/etc/asterisk/extensions.conf:
[from-sip]
exten => 200,1,Dial(SIP/${EXTEN}|300)
exten => 201,1,Dial(SIP/${EXTEN}|300)
on http://www.voip-info.org/wiki/view/Asterisk+T.38
But I don't understand how this little dialplan can make my fax go from
the HT503 to my ITSP... I understand I need to use the Dial()-command... ?!
Kind regards,
Jonas.
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Re: [asterisk-users] [ASTERISK-USER] Meetme Recording issue, recording is 2 times Faster then normal recording.
Hi, The record is not double faster, it's 50% faster (100 seconds original record -> 66.6 seconds recording). Reducing tempo by 33% without losing pitch sort of "fixes" the situation, although adds alot garbage to sound file (you can do this in Audacity). Sample rate 8kHz is OK, changing it to 5280 fixes the tempo, but reduces pitch to unacceptable. Try with more callers in a conference, does it change anything (increased/decreased tempo)? You could also try ConfBridge: http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge or other conference backends (Conference, Konference...) These could solve the problem if Dahdi is broken. On Tue, May 18, 2010 at 11:46 AM, DHAVAL INDRODIYA wrote: > Hi Motiejus, > > sorry for inconvenience , because asterisk mailing list could not accept wav > file attachment > > here i am attached a file named test.wav, > > regards > Dhaval 2010/5/18 Motiejus Jakštys : > Please check WAV headers, what is the sample rate of the file? It > should be 8kHz. Does the WAV sound normal when you decrease sample > rate by hand? > > You can just upload one WAV for testing - I'll say what may be wrong with it. > > On Tue, May 18, 2010 at 9:52 AM, DHAVAL INDRODIYA > wrote: >> hello All, >> >> i have one issue with Asterisk Meetme Application >> >> i am recording through Meetme channels through option 'r' and format for >> recording a file is 'wav' >> >> lines used is DAHDI version 2.1.0.4. and asterisk version 1.6.0.5. >> >> i have very strange problem of meetme_recording , >> >> once conference starts recording file having a recording is 2x faster than >> normal recording . >> >> is there any setting to solve it out , my card type is TE410P used E1 lines >> . >> >> please help me . any help appreciated. >> >> regards >> Dhaval >> >> >> >> >> >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.30 & T38
I have been trying to get this working with an HT-502, Asterisk 1.4.31, and
Gafachi but no luck so far.
The VSP should send a re-invite for the T.38 media change on detection of
the fax tone.
I'm using canreinvite=yes on the trunk and canreinvite=no on the HT-502
extension. Also have t38pt_udptl=yes in sip.conf and T.38 Auto & Fax
Detection Callee in the HT line settings.
On May 18, 2010 5:16 AM, "Jonas Kellens" wrote:
Hello list,
I read on voip-info.org that Asterisk 1.4 support T38 passthrough.
So I guess this means that I can have a Grandstream HT503 with T38 support
and an analogue faxmachine on the other side of my Asterisk and a
T38-account with a ITSP on the other side of my Asterisk machine, right ?!
The fax coming from the faxmachine passes the HT503 to my Asterisk and my
Asterisk sends the fax to the ITSP which places it on the PSTN.
What do I place in the dialplan ??
The only dialplan I find is :
/etc/asterisk/extensions.conf:
[from-sip]
exten => 200,1,Dial(SIP/${EXTEN}|300)
exten => 201,1,Dial(SIP/${EXTEN}|300)
on http://www.voip-info.org/wiki/view/Asterisk+T.38
But I don't understand how this little dialplan can make my fax go from the
HT503 to my ITSP... I understand I need to use the Dial()-command... ?!
Kind regards,
Jonas.
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Re: [asterisk-users] Asterisk 1.4.30 & T38
On Tue, May 18, 2010 at 6:14 AM, Jonas Kellens wrote: > I read on voip-info.org that Asterisk 1.4 support T38 passthrough. That may or may not be true. I do not know. I do know that I've had much better success with fax in 1.6 than I ever had in 1.4. My personal experience is that fax works better in 1.6. My personal prejudice is to recommend you upgrade rather than fight 1.4 for this. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] About option U in Dial Ast version 1.6.2
Has any one used this?
U(x[^arg[^...]]):
x - Name of the subroutine to execute via Gosub
arg - Arguments for the Gosub routine
Execute via Gosub the routine for the *called* channel before
connecting to the calling channel. Arguments can be specified to
the Gosub
using '^' as a delimiter. The Gosub routine can set the variable ${GO
SUB_RESULT} to specify the following actions after the Gosub returns.
${GOSUB_RESULT}:
ABORT: Hangup both legs of the call.
CONGESTION: Behave as if line congestion was
encountered.
BUSY: Behave as if a busy signal was encountered.
CONTINUE: Hangup the called party and allow the
calling party to continue dialplan execution at the next
priority.
GOTO:^^: Transfer the call
to the specified priority. Optionally, an extension, or
extension
and priority can be specified.
NOTE: You cannot use any additional action post answer options in
conjunction with this option. Also, pbx services are not run on the
peer
(called) channel, so you will not be able to set timeouts via the
TIMEOUT()
function in this routine.
Thanks
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
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[asterisk-users] Play MusicOnHold and continue with dialplan
Hi guys, Is it possible to start playing MusicOnHold to the caller but also continue with the dialplan in single extension, something like this: exten => s,1,StartPlayingMoh() exten => s,n,Wait(10) exten => s,n,Dial(someone...) exten => s,n,Wait(10) exten => s,n,Dial(someone else...) ... Regards, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Play MusicOnHold and continue with dialplan
The simplest way to do this is this: Exten => s,1,noop(dial with moh) Exten => s,n,dial(tech/1,10,m) Exten => s,n,dial(tech/2,10,m) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Sent: Tuesday, May 18, 2010 9:23 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Play MusicOnHold and continue with dialplan Hi guys, Is it possible to start playing MusicOnHold to the caller but also continue with the dialplan in single extension, something like this: exten => s,1,StartPlayingMoh() exten => s,n,Wait(10) exten => s,n,Dial(someone...) exten => s,n,Wait(10) exten => s,n,Dial(someone else...) ... Regards, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality
It happens even with just a few calls (way less than 30). I'm trying to see if Asus has something to say about this. In the meantime I'm using trunk=no and it's working fine. Thanks Vieri --- On Fri, 5/14/10, Zoa wrote: > I think that the clock resets would cause no audio or > garbled audio > every 20 minutes, not constant interference. > Could you tell us how many simultaneous calls were in the > trunk and what > the size is of 1 voice packet ? > Can you try putting maximum 30 calls per trunk (use > multiple trunks if > needed) and see if the problem goes away. > > Greetings, > > zOa > > Vieri wrote: > > --- On Thu, 5/13/10, Zoa > wrote: > > > > > >> Can you try trunk = no ? > >> > > > > Lifesaver... > > trunk=no made the "interference" go away. > > I have clean audio now. > > > > Quote: "IAX Trunking needs support of a hardware > timer." > > > > I'm supposing my system is using the DAHDI-driven > Digium cards on my motherboard. I don't know how hardware > timers work and if Digium hardware rely on the motherboard > (my system clock is going too fast and my ntpd is constantly > adjusting the clock by -2.6 seconds every 20 minutes). In > any case, since I'm on a dedicated LAN I guess I can safely > set trunk=no. > > > > Thanks! > > > > Vieri > > > > > > > > > > > > > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar > every Thurs: > > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality
- "Vieri" wrote: > It happens even with just a few calls (way less than 30). > I'm trying to see if Asus has something to say about this. > In the meantime I'm using trunk=no and it's working fine. > Have you enabled "trunktimestamps=yes"? If I recall, I was able to overcome quality issues by using that option to ensure proper timing... YMMV. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Faxes from website works, but from regular don't: cause 16
Hi Guys, I'm having a non-obvious issue, i am using Fax for asterisk to receive faxes, so when i test using a website that send faxes it's working great: the fax is received and the fax2mail app is called and i get it in my email box. but when i try using a regular fax machine everything in logs (turned on debug) but all of the sudden a line appear saying: Channel 0/1, span 1 got hangup request, cause 16 and then the fax2mail is not called for some reason and [image: :(] no fax received, can you help me guys with that? thanks!!! -- Abdullah -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Peering with a Taqua T7000
Anyone have any luck configuring a SIP trunk on a Taqua to talk to Asterisk? We were initially set up as a subscriber (access line) but that had some undesirable side-effects, such as quashing the ANI on outbound calls. Looks like we're going to have to reconfigure the trunk as a "network gateway". I asked their Director of Product Management for product documentation but didn't hear back, so I guess we're on our own. If anyone else has successfully interoperated, please share your results. Also any information about Diversion: or P-Asserted-Identity: results would also be handy. Thanks, -Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] NPA NXX Database
Has anyone had good results with an on-line database that returns a LATA based on NPA NXX? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NPA NXX Database
On May 18, 2010, at 1:13 PM, Don Kelly wrote: > Has anyone had good results with an on-line database that returns a LATA > based on NPA NXX? > --Don > > Don Kelly > There's an online list that you can convert to a locally stored db. http://www.nanpa.com/nanp1/allutlzd.zip ---fred http://qxork.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NPA NXX Database
http://www.localcallingguide.com/ will give you lots of info. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly Sent: Tuesday, May 18, 2010 12:14 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] NPA NXX Database Has anyone had good results with an on-line database that returns a LATA based on NPA NXX? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Location with PRI / Analog lines
Hi there, I am stuck with the location issues. It would be easy if you have DID for each extension so that outgoing caller id would be DID of the respective extension and also physical address. Now if you are not able to get DID's for some reason. I am thinking of some situations and appreciate your thoughts. 1. If I have a PRI and map physical number (original numbers in hunt group not the group number) to some extensions. If somebody calls emergency from a specific location in a building either it will have outgoing caller ID of that specific number in PRI group (if possible) or always dial that physical line which has address of that location in Telco database. Is it possible? Can we have different caller ID's for a PRI? I mean my PRi has one number 12345667 known to outside world. But it has 23 physical number in original.Please comment 2.If I have analog lines, incoming call can use any channel and out going calls from any extension can use any channel. If somebody dials emergency then that specific extension dials specific channel which has its physical location in Telco database. I would highly appreciate your thoughts. Shahnawaz Mir http://www.aaanetworkx.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ?
Hi Indeed, limited to only 50 BLF, thats why operator was placed two aastra phones and set a ring group for both. Best Regards Jose Flores Galicia <> BriefCode && Code Based Training 2010/5/18 Olivier > >> On Tue, May 18, 2010 at 11:58 AM, Olivier wrote: >> >>> Hi, >>> >>> Can you share successful experience with a SIP hardphone supporting 100 >>> BLF ? >>> Which phone would you suggest for that ? >>> >>> (In case that matters, each BLF is supposed to SUBSCRIBE to and reflect >>> the state (Idle, Ringing, OnCall) of a local extension. >>> >>> Regards >>> >>> >>> >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >> >> -- >> Thank you with regards, >> Gopalakrishnan A.N, >> >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] quick question on conf bridge
I have a customer that is using a quad core xeon server with 4 GIG ram and Te210P card. Currently this machine is being used for calling out to their own people as well other programs being run. anyway they wish to start using it for a 30 person conference bridge. I presume this is no issue??? I am running centos 64 and asterisk 1.4.30 I was thinking they might need to expand the card to a quad card so calling out can still take place as the 30 person conf bridge is happening. Again I presume with this machine that even using all the quad T1's isnt an issue. Is that the case? I have not done anything with conferencing on asterisk and didnt know if using the bridge is really CPU intensive or not. Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quick question on conf bridge
On Tue, 18 May 2010, Jerry Geis wrote: > I have a customer that is using a quad core xeon server with 4 GIG ram > and Te210P card. [snip] > anyway they wish to start using it for a 30 person conference bridge. I > presume this is no issue??? I am running centos 64 and asterisk 1.4.30 [snip] > I have not done anything with conferencing on asterisk and didnt know if > using the bridge is really CPU intensive or not. Assuming no aggressive transcoding, plenty of RAM, plenty of CPU. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ?
Dumb question - wouldn't it be easier to monitor a web interface than a phone with 100 lights? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jose Flores Galicia Sent: Tuesday, May 18, 2010 2:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ? Hi Indeed, limited to only 50 BLF, thats why operator was placed two aastra phones and set a ring group for both. Best Regards Jose Flores Galicia <> BriefCode && Code Based Training 2010/5/18 Olivier On Tue, May 18, 2010 at 11:58 AM, Olivier wrote: Hi, Can you share successful experience with a SIP hardphone supporting 100 BLF ? Which phone would you suggest for that ? (In case that matters, each BLF is supposed to SUBSCRIBE to and reflect the state (Idle, Ringing, OnCall) of a local extension. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you with regards, Gopalakrishnan A.N, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Play MusicOnHold and continue with dialplan
Thanks, but in my particular case I need to do pause between dials (using Wait() command). How could I implement MoH also when Wait is in progress (in single extensions that is)? Is this even possible, or is the only way to encapsulate the logic in one extension and do Dial(Local/lo...@extension,,m) in another one? Regards, Alex -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, May 18, 2010 4:31 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Play MusicOnHold and continue with dialplan The simplest way to do this is this: Exten => s,1,noop(dial with moh) Exten => s,n,dial(tech/1,10,m) Exten => s,n,dial(tech/2,10,m) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Sent: Tuesday, May 18, 2010 9:23 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Play MusicOnHold and continue with dialplan Hi guys, Is it possible to start playing MusicOnHold to the caller but also continue with the dialplan in single extension, something like this: exten => s,1,StartPlayingMoh() exten => s,n,Wait(10) exten => s,n,Dial(someone...) exten => s,n,Wait(10) exten => s,n,Dial(someone else...) ... Regards, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Play MusicOnHold and continue with dialplan
Here's one way Exten => s,1,noop(dial with moh) Exten => s,n,dial(tech/1,10,m) Exten => s,n,WaitExten(10,m) Exten => s,n,dial(tech/2,10,m) Exten => s,n,WaitExten(10,m) The waitexten(10,m) plays musiconhold waiting for a 1 digit extension. As long as there's not one in the context, you're good. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Sent: Tuesday, May 18, 2010 4:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Play MusicOnHold and continue with dialplan Thanks, but in my particular case I need to do pause between dials (using Wait() command). How could I implement MoH also when Wait is in progress (in single extensions that is)? Is this even possible, or is the only way to encapsulate the logic in one extension and do Dial(Local/lo...@extension,,m) in another one? Regards, Alex -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, May 18, 2010 4:31 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Play MusicOnHold and continue with dialplan The simplest way to do this is this: Exten => s,1,noop(dial with moh) Exten => s,n,dial(tech/1,10,m) Exten => s,n,dial(tech/2,10,m) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Sent: Tuesday, May 18, 2010 9:23 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Play MusicOnHold and continue with dialplan Hi guys, Is it possible to start playing MusicOnHold to the caller but also continue with the dialplan in single extension, something like this: exten => s,1,StartPlayingMoh() exten => s,n,Wait(10) exten => s,n,Dial(someone...) exten => s,n,Wait(10) exten => s,n,Dial(someone else...) ... Regards, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] a2billing DID and Queues
Hi all, I have configured asterisk and a2billing.for inbound i have also configured did and its forwarded to sip extensions. But i want to enable queues with inbound numbers(DID).But i could not find a way to do this in a2billing. I want enable that if some did comes to asterisk/a2billing it should be forwarded to queues not sip extensions and their i want to enable hunting so if one extensions does not receive the call so it should be forwarded to the next extensions. So please help, Any help will highly appreciated. Thanks -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Cluster
Hello Everyone, I must deploy an asterisk system that can support at least 500 pstn outbound calls. It's a challenge as it's the first time i'm gonna build such a large system. I want to have your advice on hardware, software and so on . What i have in my plan is a cluster of servers with quad PRI cards. I will appreciate your advice. Thank you all . -- Adolphe CHER-AIME Network Integrator CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3748-3875 / (509) 3449-4280 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing DID and Queues
Hello
I think you can do this using Local Channel
for example I have do so:
queues.conf
[MyQueue]
musicclass = default
ringinuse = yes
strategy=leastrecent
joinempty = yes
timeout=60
retry=5
weight=0
wrapuptime=1
maxlen = 0
announce-frequency = 10
announce-holdtime = no
periodic-announce = knereq_bolor_oper_zbaxvac_zang_poqrush
periodic-announce-frequency = 30
announce-round-seconds = 10
reportholdtime = no
timeoutrestart = no
member => Local/5...@callcenter/n
member => Local/5...@callcenter/n
context = callcenter
context callcenter {
_500. => {
Noop(${EXTEN});
Noop(${incpeerunique});
Noop(${CALLERID(all)});
operphone="phone${EXTEN:3:1}";
Noop(Call count:${SIPPEER(phone1:curcalls)});
DB(${operphone}/inccid)=${CALLERID(number)};
SetMusicOnHold(default);
Dial(SIP/${operphone},,tTg);
Noop(5001);
};
h => {
//NoCDR;
Noop(Hangup in callcenter context);
DB(${operphone}/inccid)=""};
};
};
context a2bdid {
_X. => {
Noop(${CALLERID(rdnis)});
Noop(${CALLERID(number)});
Noop(${CALLERID(name)});
Noop(${CALLERID(all)});
Set(CHANNEL(language)=am) ;
Noop(${QUEUE_WAITING_COUNT(MyQueue)});
Noop(${QUEUE_MEMBER_COUNT(MyQueue)});
Ringing;
Queue(MyQueue,tTr);
Noop(Posle Queue);
Noop(Vau);
Hangup;
};
h => {
Noop(Hangup in callcenter1 context);
};
};
This is work for me.
Best regards,
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
toqeer ali wrote:
> Hi all,
>
> I have configured asterisk and a2billing.for inbound i have also
> configured did and its forwarded to sip extensions.
>
> But i want to enable queues with inbound numbers(DID).But i could not
> find a way to do this in a2billing.
>
>
> I want enable that if some did comes to asterisk/a2billing it should be
> forwarded to queues not sip extensions and
>
> their i want to enable hunting so if one extensions does not receive the
> call so it should be forwarded to the next
>
> extensions.
>
> So please help, Any help will highly appreciated.
>
> Thanks
>
> --
> Toqeer Ali Syed
>
> Red Hat Certified Engineer
> mob: +92 321 9059916
>
--
_
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Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ?
2010/5/18 Danny Nicholas > Dumb question – wouldn’t it be easier to monitor a web interface than a > phone with 100 lights? > Yes and no : operator already has a Flash Operator Panel on its screen. Information displayed by FOP is richer (you can see who is talking to who) but operator feels easier with dedicated buttons for both displaying activity and issuing transfers. I think 100 is the upper limit for both kinds of tools where at a glance, you can see all extensions : I think above a certain user count (120 ?), operator would prefer to specifically query its console to get current specific extensions phone activity. > > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jose Flores > Galicia > *Sent:* Tuesday, May 18, 2010 2:32 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ? > > > > Hi > > > > Indeed, limited to only 50 BLF, thats why operator was placed two aastra > phones and set a ring group for both. > > > > Best Regards > Jose Flores Galicia > <> > BriefCode && Code Based Training > > 2010/5/18 Olivier > > > > On Tue, May 18, 2010 at 11:58 AM, Olivier wrote: > > Hi, > > Can you share successful experience with a SIP hardphone supporting 100 BLF > ? > Which phone would you suggest for that ? > > (In case that matters, each BLF is supposed to SUBSCRIBE to and reflect the > state (Idle, Ringing, OnCall) of a local extension. > > Regards > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > Thank you with regards, > Gopalakrishnan A.N, > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [ASTERISK-USER] Meetme Recording issue, recording is 2 times Faster then normal recording.
hi Motiejus, Can you give a command for converting it to normal voice , in audacity. also i tired with more users still problem persists , can i try with gsm format , what you say? regards Dhaval 2010/5/18 Motiejus Jakštys > Hi, > The record is not double faster, it's 50% faster (100 seconds original > record -> 66.6 seconds recording). Reducing tempo by 33% without > losing pitch sort of "fixes" the situation, although adds alot garbage > to sound file (you can do this in Audacity). > Sample rate 8kHz is OK, changing it to 5280 fixes the tempo, but > reduces pitch to unacceptable. > > Try with more callers in a conference, does it change anything > (increased/decreased tempo)? > > You could also try ConfBridge: > http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge > or other conference backends (Conference, Konference...) > These could solve the problem if Dahdi is broken. > > On Tue, May 18, 2010 at 11:46 AM, DHAVAL INDRODIYA > wrote: > > Hi Motiejus, > > > > sorry for inconvenience , because asterisk mailing list could not accept > wav > > file attachment > > > > here i am attached a file named test.wav, > > > > regards > > Dhaval > > > > 2010/5/18 Motiejus Jakštys : > > Please check WAV headers, what is the sample rate of the file? It > > should be 8kHz. Does the WAV sound normal when you decrease sample > > rate by hand? > > > > You can just upload one WAV for testing - I'll say what may be wrong with > it. > > > > On Tue, May 18, 2010 at 9:52 AM, DHAVAL INDRODIYA > > wrote: > >> hello All, > >> > >> i have one issue with Asterisk Meetme Application > >> > >> i am recording through Meetme channels through option 'r' and format for > >> recording a file is 'wav' > >> > >> lines used is DAHDI version 2.1.0.4. and asterisk version 1.6.0.5. > >> > >> i have very strange problem of meetme_recording , > >> > >> once conference starts recording file having a recording is 2x faster > than > >> normal recording . > >> > >> is there any setting to solve it out , my card type is TE410P used E1 > lines > >> . > >> > >> please help me . any help appreciated. > >> > >> regards > >> Dhaval > >> > >> > >> > >> > >> > >> > >> > >> -- > >> _ > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >> New to Asterisk? Join us for a live introductory webinar every Thurs: > >> http://www.asterisk.org/hello > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ?
On Wed, 19 May 2010, Olivier wrote: > 2010/5/18 Danny Nicholas > >> Dumb question ? wouldn?t it be easier to monitor a web interface than a >> phone with 100 lights? >> > Yes and no : operator already has a Flash Operator Panel on its screen. > Information displayed by FOP is richer (you can see who is talking to who) > but operator feels easier with dedicated buttons for both displaying > activity and issuing transfers. I've deployed a few Grandstream phones with a single button-box. The box has 56 additional keys to the 7 on the GXP2000. It can support a 2nd box connected in to give 119 BLF buttons/speed-dials. BLF was flakey in early Grandstreams, but seems to work fine for me in these systems - but I know Grandstream gets a bad rap by many these days, but it's not too expensive for a trial. Gordon > > I think 100 is the upper limit for both kinds of tools where at a glance, > you can see all extensions : I think above a certain user count (120 ?), > operator would prefer to specifically query its console to get current > specific extensions phone activity. > >> >> -- >> >> *From:* asterisk-users-boun...@lists.digium.com [mailto: >> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jose Flores >> Galicia >> *Sent:* Tuesday, May 18, 2010 2:32 PM >> *To:* Asterisk Users Mailing List - Non-Commercial Discussion >> *Subject:* Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ? >> >> >> >> Hi >> >> >> >> Indeed, limited to only 50 BLF, thats why operator was placed two aastra >> phones and set a ring group for both. >> >> >> >> Best Regards >> Jose Flores Galicia >> <> >> BriefCode && Code Based Training >> >> 2010/5/18 Olivier >> >> >> >> On Tue, May 18, 2010 at 11:58 AM, Olivier wrote: >> >> Hi, >> >> Can you share successful experience with a SIP hardphone supporting 100 BLF >> ? >> Which phone would you suggest for that ? >> >> (In case that matters, each BLF is supposed to SUBSCRIBE to and reflect the >> state (Idle, Ringing, OnCall) of a local extension. >> >> Regards >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> >> -- >> Thank you with regards, >> Gopalakrishnan A.N, >> >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
