Displaying 20 results from an estimated 10000 matches similar to: "Sending Re-Invite with Dialplan application?"
2009 Apr 30
3
need help on asterisk call forwarding
Hello,
I am trying to enable call forwarding feature on asterisk 1.6.0.9 with
asterisk-gui. Sure there is no menu for that on gui but, when i try to
write some example scripts to extensions.conf to make it work. I totally
failed.
I dont wanna install smthing like freepbx etc on the system so, i need
help to add unconditional etc call forwarding feature for 1.6
Thanks
2008 Oct 23
0
command - set sip_codec- does not work with asterisk-1.4.21
hello:
i want to test the g729 with asterisk. my scenario is sipp(ulaw)->asterisk1 with g729->asterisk2 with g729.
I want to test g729 module with asterisk-1.4.21, when i make calls from asterisk 1 to asterisk 2, the asterisk 1 always send ulaw to asterisk 2. my sip in asterisk 1 is with codec g729 and enforce that use g729, the sip in asterisk 2 also work with G729 only, but asterisk 2
2009 Feb 25
1
SIP_CODEC variable
Hi,
I am using Aserisk 1.4.23.1 and trying to use SIP_CODEC to define the codec
being used. I have exclusively Polycom phones for this test, and basically I
want all communications to use g729 (preferred codec), except for pagine 20
phones (which busts my g729 license count). In that case I want to use gsm.
I have therefore specified Set(SIP_CODEC=gsm) I my dialplan before the
appropriate
2003 May 05
3
G723 - Has anyone gotten SIP_CODEC= to work?
FYI, asterisk DOES now support g723, but you have to pay for it:
http://store.yahoo.com/asteriskpbx/asteriskg729.html
-----Original Message-----
From: Dan Fernandez <danfernandez00@hotmail.com>
Date: Mon, 5 May 2003 17:33:05 -0300
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work?
Basically, since I?d like to use g723 for sip
2004 Jun 24
2
How to force G729
We want some of our users to use G729, and some others to use ULAW. Our PSTN gateway provider supports both, so that's not a problem, and if I force him (the PSTN gateway) to allow G729 only, the outgoing call will take place with G729.
The problem is that I want to have my PSTN provider configured to allow ULAW as a first priority, then G729. I did it like that:
[mypstngate]
type=friend
2005 Jun 03
0
SIP_CODEC, reinvites, and changing codecs
I am wondering if the SIP protocol and its implementation in * allows for
changing codecs mid-connection.
I've seen some questions regarding this on the list, but I've not found any
clear answers.
I've also seen the SIP_CODEC variable, but it's not clear that it will change
the codec on an existing call. Also, there are mentions of needing a reinvite
to make the change, but most
2013 Apr 10
1
Regarding Uploading & Editing & Saving Docments In a Rails Application !
Hi,
i am working on Application !
I am uploading Word Document with *Paperclip* to the application.File is
saved in *Public/.
I want edit that document in a Rails Application.*I want save it back to
application.
Is there chance to do this ?
What are gems are helpful for this *integration*.
Thanks,
Sai
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You received this message because you are subscribed to the Google Groups "Ruby
2020 Sep 25
0
Negotiates g729 but RTP contains g711
Hi,
I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing
2020 Sep 25
0
PJSIP - Forcing codec preference?
Hi,
We're holding ourselves back from moving to PJSIP as we don't appear to have figured out how to force codec preference in a dial plan. The 'PJSIP Advanced Codec Negotiation' document (https://wiki.asterisk.org/wiki/display/AST/PJSIP+Advanced+Codec+Negotiation) appears to ultimately be what we're after, but we're not comfortable running Asterisk 18 in production just
2006 Dec 19
0
dtmf and ivr
hello,
i try to build a IVR for our company my problem is that the dtmf tones
are not recognized by the phones i tried several phones.
BUT when i call the voicemail i can navigate with all phones through the
menu. I use * 1.2
here is the context:
[ivr]
exten => s,1,Answer
exten => s,2,SetMusicOnHold(default)
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,10
;SAI menu -
2014 Sep 23
1
Change codec when dial from SIP to DAHDI
Hi:
I am useing asterisk 11.12.
I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI
use alaw. G722 is great when ip-phone talks to each other. but when
ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to
transcode to alaw.
so I try to change the codec when dial from SIP to DAHDI. I tried
to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP
2017 Jun 08
2
help regarding r-project
hello sir/ma'am,
trying to build a small prototype of resume filter
but not able to get expected intercept values, could you please help
regarding this project ASAP.
to predict, skills + experience =result
--
*With Regards,*
*C Sai Sathvick*
1999 Jan 24
0
Re: util-linux compromised
I just received the following letter:
Date: Sun, 24 Jan 1999 04:01:55 -0500 (EST)
From: John Stange <building@cs.umd.edu>
Subject: util-linux compromised?
I grabbed util-linux-2.9g yesterday from win.tue.nl, and discovered a
section of login.c that appears to send the host and uid of the user to a
hotmail address. I imagine this isn't a standard feature. :>
2014 Jul 30
2
SIP trunk gives fuzzy / distorted audio on mobiles, OK on fixed lines
I'm having a problem with a new SIP trunk.
Calls within the UK to fixed lines are fine, but calls to mobiles have
noticeably poorer audio quality.
I thought it might have been a codec issue; we have used G.726 for internal
and external calls (over primary ISDN and GSM). So I tried allowing "alaw",
(G.711 A-law) which is the native codec used within the PSTN in this country,
2004 Mar 29
2
cut and factor
Eric,
thanks for quick reply. at first look I thought it is what I need,
but, unfortunately, it doesn't applied to original data - it
creates new data with loosing original indexes ! I want to keep indexes
of original data, but replace original data with $mids of corresponding
$breaks.
So, if I have z = 1:10, t=hist(z,plot=F)
> z
[1] 1 2 3 4 5 6 7 8 9 10
> t$breaks
[1] 0 2
2020 Sep 25
0
Negotiates g729 but RTP contains g711
Hi,
I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing
2003 Oct 20
1
Setvar SIP_CODEC
Hello,
I have
a couple of 7960 and a quad T1 card on my asterisk box. I want to let
the phones to use g729 when they "talk" to each other, but to use g711
when I'm going to route the call out of my network using the T1 card.
Everything works just fine between the phones, but in order to be able
to make calls through T1 I have to disallow the g729.
For this purpose I have the
2017 Jun 08
0
help regarding r-project
I do not think anyone is going to be able to help you unless you can
provide a reproducible example with a clear account of what it dies and
what you expected it to do.
On 08/06/2017 06:34, Sai Sathvick wrote:
> hello sir/ma'am,
>
> trying to build a small prototype of resume filter
> but not able to get expected intercept values, could you please help
2014 Sep 27
2
can PJSIP_MEDIA_OFFER work like SIP_CODEC?
hi:
when using chan_sip, I can use set SIP_CODEC in dialplan to change
the codec of endpoint. this method didn't work with pjsip in asterisk
12/13.
I found asterisk 12/13 has a new function PJSIP_MEDIA_OFFER.
according to the description, it seems can set codec, but the document
didn't offer any example. i try to use something like
PJSIP_MEDIA_OFFER(alaw) but didn't work.
2009 Jul 22
0
Attended transfer and 'pbx-invalid' - 1.4.26
Hi,
I've created a tiny dialplan to test the return of a call on transfers,
like this: (I had to use the DEVSTATE backport here)
[phones]
exten => _12XX,1,Dial(SIP/${EXTEN},6,tT)
exten => _12XX,n,GotoIf($[ "x${BLINDTRANSFER}" = "x" ]?noBT)
exten => _12XX,n,Set(DIALRET=${CUT(BLINDTRANSFER,-,1)});
exten => _12XX,n,Goto(dRet)
exten => _12XX,n(noBT),GotoIf($[