Hi Guys
Given these occassional errors on my Asterisk CLI:
[Jul 2 10:23:36] WARNING[2060]: chan_sip.c:3641 retrans_pkt: Retransmission
timeout reached on transmission
17bb3a993ad10f8818970ae952b81e73@192.168.11.31:5060 for seqno 102 (Critical
Request) -- See
being honest with i have been lost on what to do.
all i want is sent from my email a pdf file and then the server will sent it as fax.
what settings do i have to do regarding emailing to the server? what other settings do i have to do?
is there a guide on that?
Sent:Friday, June 26, 2015 at
On Wed, Jun 24, 2015 at 7:03 AM, Jerry Geis ge...@pagestation.com wrote:
I am looking for some great instructions on using asterisk
with pulse.
I'm using centos 7 and pulse as a user and not having much luck.
I have changed all permissions for the asterisk directories.
set asterisk.conf
On Wed, Jul 1, 2015 at 4:46 AM, r...@yandex.ru wrote:
Hi, all
Is there someway ability to insert custom Header to SIP 486 message,
when HANGUP application is invoked?
Our use case is to set that Header, when call-limit is reached, to analyze
elsewhere, but we do not want to set some custom
Is there any chance to create feature request for that useful functionality? 02.07.2015, 14:03, "Rusty Newton" rnew...@digium.com:On Wed, Jul 1, 2015 at 4:46 AM, r...@yandex.ru wrote:Hi, all Is there someway ability to insert custom Header to "SIP 486" message, when HANGUP application is invoked?
Thanks for the tip. Our goal is to know that call-limit is triggered. And later analyze this
info, maybe do some action.
Yes, we can parse CDRs or execute AGI script but we do not want inmplement this logic on
Asterisk because it can affect performance.
02.07.2015, 15:31, jg
Thanks for the tip. Our goal is to know that call-limit is triggered. And later analyze this info, maybe do some action.Yes, we can parse CDRs or execute AGI script but we do not want inmplement this logic on Asterisk because it can affect performance. 02.07.2015, 15:31, "jg"
Is there any chance to create feature request for that useful functionality?
02.07.2015, 14:03, Rusty Newton rnew...@digium.com:
On Wed, Jul 1, 2015 at 4:46 AM, r...@yandex.ru mailto:r...@yandex.ru wrote:
Hi, all
Is there someway ability to insert custom Header to SIP 486 message,
* call-limit on PBX is triggered 02.07.2015, 15:49, "r...@yandex.ru" r...@yandex.ru:Thanks for the tip. Our goal is to know that call-limit is triggered. And later analyze this info, maybe do some action.Yes, we can parse CDRs or execute AGI script but we do not want inmplement this logic on
being honest with i have been lost on what to do.
all i want is sent from my email a pdf file and then the server will sent it as
fax.
what settings do i have to do regarding emailing to the server? what other settings do i have
to do?
is there a guide on that?
*Sent:* Friday, June 26, 2015
At the moment it receives fax and sends it as pdf to my email.
i have followed this tutorial http://the-asterisk-book.com/1.6/faxserver-mit-iaxmodem-und-hylafax.html
i have setup postfix to send using my gmail account.
how do i do the reverse to send from my email?
Sent:Thursday, July 02,
HI LIST CAN U HELP ME
If there are multiple sip trunks with the same ITSP then an incoming call
is arbitarily matched to the last peer with the same host IP address. This
is not a serious problem because the DID is still correct but it does have
many insidious effects due to the incorrect channel
I'm not sure that your question is clear. You'll probably want to be more
specific.
What is pulse? You mention as a user, are you talking about voicepulse.com
?
What are you trying to do with pulse?
What problem are you running into?
Sorry Rusty...
I am trying to get Asterisk 11 to
I am trying to get Asterisk 11 to co-exist with a centos 7 box that has pulse
audio running as a local user.
Has anyone done that?
What is the trick?
I changed directories /var/run/asterisk, /var/spool/asterisk,
/var/log/asterisk,
/usr/lib/asterisk, /etc/dahdi and all that stuff.
I use both confbridge to bring several devices into
a receive only or listen mode, then allow the one person
on the phone to speak live over those devices.
Works great.
However - now I would like to be able to play a tone
into the conference before the person speaks.
How might that be
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