[asterisk-users] is g729 codec free? or under license???
hello all, i have problem in using g729 codec. my asterisk version is 1.8.22. when i run "core show codecs" in asterisk, there is a g729 codec in the list so i assume that i can use it for my channels. but connection can not be set when i use it for my h323 channel. i read somewhere that codec g729 is a commercial codec and i should buy its license in order to use it. is it true? if yes, why is it listed in codecs in asterisk?? thanks in advance, SAM -- _ -- 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] Direct DAHDI documentation
Hello, I wanted to switch from using Dialogic/Eicon cards to using Digium's T-1 cards. When I purchased a sample card the salesperson assured me there was documentation specific to the DAHDI interface. Now that I'm digging in, I'm finding it's documented a lot like Linux -- one must read the fairly uncommented source code. I don't have a problem with this generally, but here I just don't understand the divisions of labor between Asterisk, DAHDI Hardware, DAHDI kernel modules and Userland (me). (BTW, I do not wish to use Asterisk as we have numerous projects based on Dialogic/Eicon spanning some 20 years. My intent is to write a replacement look-a-like driver which uses Digium's cards instead of Dialogic's.) My specific issues are: 1) HDLC. Does the hardware have an HDLC controller, or is it the user's job to hunt for flags, frame the data and calc the FCS? 2) ISDN/PRI. Does the kernel module load Q.921/931 implementation or is this user's responsibility? I know there's a LIBPRI product, which I may use, but I have my own PRI library which was confirmance tested with ATT years ago. Either way, I'm not sure how the D-channel data is flowing. 3) I got the idea that B-channel data is collected by the kernel module in 8 sample blocks (1 ms). Does this mean I need to be reading it out/writing it in at that rate? I saw some buffering code, but wasn't sure if that was voicefile type playback/record or if all audio is treated without regard to its source/destination. I guess I could lock onto it at 1ms using Linux's HPET timer, although that sounds clumsy. 4) I can certainly convert between ulaw/linear to sum for conferencing, but it seems the kernel module might support that as well? Or at the least it seems the kernel module can support chan-to-chan connections. 5) I found some DTMF (FIR goertzel) code somewhere in DAHDI, but also in Asterisk. While I have such code in own library, am I to understand DTMF can be detected within the kernel module? I guess I really would like to see a doc on the overall concept of DAHDI hardware and its kernel module. I don't care how it's laid out, I'd just like to get my mind around it. Does anyone know of an example telephony C file that might show: 1) initialization of DAHDI spans 2) waiting for inbound events 3) answering a call 4) sending a voice file, recording a voice file 5) disconnection of calls 6) de-initialization And perhaps showing how two channels are connected to create a conversation? Thanks in advance, Gary -- _ -- 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] SIM adaptor (huwewi or other)
Hi, We've used https://code.google.com/p/asterisk-chan-dongle/ in the past with success, only one call per sim On 29 September 2013 09:39, bilal ghayyad wrote: > > > > On Wednesday, September 11, 2013 1:54 PM, longst > wrote: > I think GoIP gsm gateway also is a good choice > > Sent from Shitian Long > > > On Sep 11, 2013, at 12:29 PM, bilal ghayyad wrote: > > Hello; > > I am looking for SIM adaptor to be connected with Asterisk to be able to > send and receive calls from the mobile operator and if possible the same > adapter to be used for SMS "sending and receiving". > > But what if anyone called this SIM card that is connected to this adapter > and no one relied his call, how this miss call can reach for the use at the > asterisk PBX? > > Regards > Bilal > > -- > _ > -- 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] confbridge - play different sounds to caller and bridge at same time?
When a caller enters the confbridge, I want to play a sound file ('ring')
for the caller and a different sound file ('type of caller') to the bridge
(all participants or just the admin?) at the same time.
It's OK if the bridge hears the 'ring,' but the caller should not hear the
'type of caller.'
Any clues?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] iax: unable to transfer - one way audio
On 09/28/2013 11:11 AM, Asghar Mohammad wrote:
Hi,
If you post your configuration someone may help you.
On Sat, Sep 28, 2013 at 5:03 PM, Sean Darcy mailto:seandar...@gmail.com>> wrote:
On 09/27/2013 09:08 PM, Sean Darcy wrote:
We have zoiper connected over iax to asterisk in Sydney. The
call is to
asterisk in New York. The caller in NZ can hear clearly. Nothing
in NY.
Here's the sydney server:
-- Accepting AUTHENTICATED call from :
> requested format = speex,
> requested prefs = (),
> actual format = ulaw,
> host prefs = (silk16|ulaw|gsm|g722),
> priority = mine
-- Executing [8447@nz-in:1] Dial("IAX2/n4-270",
"IAX2/sydney") in
new stack
-- Called IAX2/sydney
-- Call accepted by (format ulaw)
-- Format for call is (ulaw)
-- IAX2/sydney-8819 is ringing
-- IAX2/sydney-8819 answered IAX2/n4-270
-- Channel 'IAX2/n4-270' unable to transfer
-- Channel 'IAX2/sydney-8819' unable to transfer
-- Channel 'IAX2/sydney-8819' unable to transfer
-- Channel 'IAX2/sydney-8819' unable to transfer
The NY server:
-- Accepting AUTHENTICATED call from :
--> requested format = ulaw,
--> requested prefs = (ulaw|silk16|gsm|g722),
--> actual format = ulaw,
--> host prefs = (ulaw|gsm|g722),
--> priority = mine
-- Executing [s@incoming-nz:1] Goto("IAX2/home-2152",
"incoming,s,nz-in") in new stack
-- Goto (incoming,s,5)
-- Executing [s@incoming:5] Dial("IAX2/home-2152",
"DAHDI/g0&SIP/250&SIP/251,60,__tT") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called DAHDI/g0
-- Called SIP/250
-- Called SIP/251
-- DAHDI/1-1 is ringing
-- SIP/251-001d is ringing
-- SIP/250-001c is ringing
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 answered IAX2/home-2152
-- Channel 'IAX2/home-2152' unable to transfer
-- Hanging up on 'DAHDI/1-1'
Any help appreciated.
sean
FWIW, sydney server is 11.5.1, ny server 11.6.0-rc1.
sean
Thanks for the reply.
Here's sydney iax.conf:
[general]
bandwidth=medium
trunkmtu=1240
disallow=all
allow=silk16
allow=ulaw
allow=gsm
allow=g722
jitterbuffer=yes
forcejitterbuffer=no
trunktimestamps=yes
authdebug=yes
tos=ef
cos=5
autokill=yes
codecpriority=caller
[default](!)
type=friend
auth=md5
host=dynamic
context=nz-in
qualify=1000
setvar=Protocol=IAX2
[n4](default)
secret=
callerid=""
[sydney](default)
secret=
username=home-sydney
home iax.conf:
[general]
bandwidth=medium
disallow=all
allow=ulaw
allow=gsm
allow=g722
jitterbuffer=yes
forcejitterbuffer=no
tos=0x10
autokill=yes
register => sydney:@
[nz](!)
type=friend
secret=
context=incoming-nz
[home-sydney](nz)
host=
username=sydney
callerid="House"
sean
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Re: [asterisk-users] problem to get MWI working
Thanks a lot the logs were handy. I thought just activating it through the CLI would be enough! The script was executed correctly, but couldn't see the results because it was running in a different shell. Once I fixed it, I started to see some output! -- _ -- 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] QueueWiz - a free call-center simulator tool for Asterisk
Hello all, next week it's Astricon 10 time, so we thought we'd create something that the community could like and use for free. It's a pretty effective tool if you run a call-center or plan to run one. QueueWiz is the first free web app for interactive, quick and accurate call center sizing, cost and revenue simulation. Insert your data with the intuitive interface, measure traffic intensity, expected wait times, agents' engagement, revenue per call and per agent and even hourly margins. Save your simulation and share it via email or social media. Completely free of charge - no string attached - try it at http://queuewiz.queuemetrics.com Have a great day and see you next week. l. -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.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
