Hello,
I'm having this issue on my pbx, it appears that asterisk is refusing
the codecs that my providers is proposing.
My trunk configuration is:
---
username=5x5x7x9x0x3
type=friend
secret=CRcxn7sqwm
qualify=yes
port=5060
insecure=port,invite
host=sip.txtxlxoxp.it
fromuser=5x5x7x9x0x3
On 1/15/2014 3:59 AM, Francesco Namuri wrote:
Hello,
I'm having this issue on my pbx, it appears that asterisk is refusing
the codecs that my providers is proposing.
My trunk configuration is:
Pretty simple -
---
username=5x5x7x9x0x3
type=friend
secret=CRcxn7sqwm
qualify=yes
port=5060
I could replace the card this morning and the timing slips disappeared.
Given Adrian's testimony, that doesn't mean the card is faulty but as the
card is now off service, I'm really eagger to investigate further.
At the moment, I can't insert this card and test it again in my lab but
I'll
Il 15/01/2014 10.09, James Sharp ha scritto:
On 1/15/2014 3:59 AM, Francesco Namuri wrote:
Hello,
I'm having this issue on my pbx, it appears that asterisk is refusing
the codecs that my providers is proposing.
My trunk configuration is:
Pretty simple -
---
username=5x5x7x9x0x3
On 15/01/14 09:39, Francesco Namuri wrote:
Hello James,
thanks for your answer, I supposed this too, but my provider answered me
that as
m=audio 43718 RTP/AVP 8 18 3 101
^ ^ ^ GSM proposal
^
On 1/15/2014 5:50 AM, Gareth Blades wrote:
On 15/01/14 09:39, Francesco Namuri wrote:
Hello James,
thanks for your answer, I supposed this too, but my provider answered me
that as
m=audio 43718 RTP/AVP 8 18 3 101
^ ^ ^ GSM proposal
Hello,
My target system is :
PSTN --- Sip Provider ---IP/ADSL--- Router with fw/NAT --- SIP/IP/Eth
-- Asterisk --- SIP/IP/Eth -- SIP Phones
Asterisk is configured to keep NAT connection with SIP provider open (with
qualifyfreq) so I don't have any problem (yet) with either casual incoming
or
Il 15/01/2014 09.59, Francesco Namuri ha scritto:
Hello,
I'm having this issue on my pbx, it appears that asterisk is refusing
the codecs that my providers is proposing.
My trunk configuration is:
---
username=5x5x7x9x0x3
type=friend
secret=CRcxn7sqwm
qualify=yes
port=5060
There is a step-by-step tutorial with video on the Digium blog:
http://blogs.digium.com/2012/11/14/how-to-install-asterisk-11-on-ubuntu-12-4-lts/
Additionally Emiliano's advice is excellent
You can read Asterisk - The future of telephony and get a lot of stuff.
Emiliano.
However, I would
El 15/01/14 12:21, Billy Chia escribió:
However, I would recommend you read Asterisk the Definitive Guide.
The Future of Telephony is now an outdated version of the book and the
name has been changed to the Definitive Guide. In the modern version
of the book there is installation instruction
On Wed, Jan 15, 2014 at 04:00:19AM +, Rodrigo Borges Pereira wrote:
Ok, I'm a believer now:
shell chrt -f 99 patlooptest /dev/dahdi/1 -t 300 -v
Using Timeout of 300 Seconds
Going for it...
Timeout achieved Ending Program
Test ran 28295 loops of 2039 bytes/loop with 0 errors
Take a look at http://www.ispeech.org/
I implemented Speech-Recognition. The API is well documented and easy.
Am 10.01.2014 21:16, schrieb Jai Rangi:
Hello,
Anyone know good quality text to speach engine for building IVRs for
asterisk. Open-source will be nice, but I wont mind paying for
Still no luck:
shell pgrep asterisk
shell ps -ef | grep asterisk
shell lsof /dev/dahdi/channel
shell lsof /dev/dahdi/1
shell lsof /dev/dahdi/2
shell chrt -f 99 patlooptest /dev/dahdi/2 -t 300 -v
/dev/dahdi/2: Device or resource busy
I can assure no PBX related process is running.
On Wed,
I tried it on 123. Those user and PW did not work :-(
Sent from my Verizon Wireless 4G LTE DROID
Shaun Ruffell sruff...@digium.com wrote:
On Wed, Jan 15, 2014 at 04:00:19AM +, Rodrigo Borges Pereira wrote:
Ok, I'm a believer now:
shell chrt -f 99 patlooptest /dev/dahdi/1 -t 300 -v
On Wed, Jan 15, 2014 at 04:18:55PM +, Rodrigo Borges Pereira wrote:
Still no luck:
shell pgrep asterisk
shell ps -ef | grep asterisk
shell lsof /dev/dahdi/channel
shell lsof /dev/dahdi/1
shell lsof /dev/dahdi/2
shell chrt -f 99 patlooptest /dev/dahdi/2 -t 300 -v
/dev/dahdi/2:
No, I see this:
shell cat /proc/dahdi/2
Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 (MASTER) B8ZS/ESF LOOP
CRC4 error count: 1
E-bit error count: 1
25 TE2/0/2/1 Clear Master LOOP
26 TE2/0/2/2 Clear LOOP
27 TE2/0/2/3 Clear LOOP
28 TE2/0/2/4 Clear LOOP
29 TE2/0/2/5 Clear LOOP
30
On Wed, Jan 15, 2014 at 05:02:10PM +, Rodrigo Borges Pereira wrote:
No, I see this:
[snip]
shell cat /proc/dahdi/1
Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 B8ZS/ESF LOOP
1 TE2/0/1/1 Clear Master RED
2 TE2/0/1/2 Clear RED
3 TE2/0/1/3 Clear RED
4 TE2/0/1/4 Clear RED
Shaun, sorry to insist on this, but this kind of diagnostic is new for me
and extremely valuable for my support operations.
You say when configured this way.. do you mean regarding dahdi? Here's my
system conf, based on your KB article for loop testing:
shell cat system.conf
loadzone=us
On Wed, Jan 15, 2014 at 05:29:44PM +, Rodrigo Borges Pereira wrote:
Shaun, sorry to insist on this, but this kind of diagnostic is new for me
and extremely valuable for my support operations.
You say when configured this way.. do you mean regarding dahdi? Here's my
system conf, based on
On Wed, 15 Jan 2014, Patrick Lists wrote:
Would you mind sharing where you get the per country IP ranges from?
I confess I 'brute forced' it by entering '/8s' into ARIN's web page and
noting if the block had been assigned to a 'foreign' NIC -- not really a
reliable and robust methodology,
Got it! It's working fine with 25. Actually, I just revisited your KB
article and it does clearly instruct to test 1, 25, and so forth... But I
was still thinking spans, not channels. So my apologies for this RTFM
moment.
Thanks!
On Wed, Jan 15, 2014 at 5:43 PM, Shaun Ruffell
Hi Steve,
On 15-01-14 18:53, Steve Edwards wrote:
On Wed, 15 Jan 2014, Patrick Lists wrote:
Would you mind sharing where you get the per country IP ranges from?
I confess I 'brute forced' it by entering '/8s' into ARIN's web page and
noting if the block had been assigned to a 'foreign' NIC
Hello,
I have an asterisk box with a peer configured with nat=force_rport,comedia,
but asterisk keeps sending the audio to the private IP address and ignoring
the client peer nat settings.
If I check the sip show peer extension, I see both symmetric RTP and
Force Rport are set to yes, but
On 1/15/14, 6:11 PM, Leandro Dardini wrote:
Hello,
I have an asterisk box with a peer configured with
nat=force_rport,comedia, but asterisk keeps sending the audio to the
private IP address and ignoring the client peer nat settings.
Why don't you try with nat=yes. It should be equivalent to
Here is list of top multi language TTS engines
1. Acapela
2. Ivona
3. Loguendo
4. Cepstral.
As per my information, they all work with open source Asterisk however
please contact with their support for more information
Regards
*Tahir Almas*
Managing Partner
ICT Innovations
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