Hi !
I have an application where I originate a call with a call file and play
some pre-recorded message when the person answers. And it's working
correctly.
Now, I've been asked to add the support for extenstion numbers.
I've been able to actualy send the extension numbver via the SendDTMF
Hi all !
We currently have an asterisk box that is rather old (runs Asterisk
1.4.21.2), and it's connected to the PSTN with a sangoma A104d card.
Now we have a new PRI at another location, and I use that occasion to
build 2 new servers, one to replace our aging one and a new one for this new
(...)
My guess is your new setup is trying to do a PRI 2B Transfer (meaning
that Asterisk is trying to handoff two B channels of a PRI to the
upstream switch). It is probably being rejected and the call is hanging
up. You will need to dig into the PRI debug of both scenarios and
compare.
In asterisk CLI do pri show spans. The fact the card is in RED alert
means the hardware does not see the pri line connected to the card.
I probably made a mistake in copying / pasting. pri show spans was showing
something like :
PRI span 1/0: Provisioned, Up, Active
Calls can enter, I see
Show us the output of a failed call with pri debug enabled on that span.
It will be difficult, since the PRI is in use on our old asterisk box.
I will have to get to the colo at night, to avoid disrupting calls
during the day.
Is there any other thing that I should collect ?
--
Le 2011-05-09 09:31, Jim Dickenson a écrit :
Make sure the firmware on the card is latest. I had a problem, not like your,
and flashing the card to the latest firmware resolved it.
It appears it did not change anything...
So, to re-cap, I have a sangoma A101 card, with the firmware uptodate,
Hi !
We curently have a centos 5 / asterisk 1.4 server that we have some DTMF
problems with. It has a Sangoma A104d card and only port one is used to
connect to the PSTN. Port 2 is conencted via a cross-over cable to a RAS for
modem access and port 3 is connected for data communication via
Make sure the firmware on the card is latest. I had a problem, not like
your, and flashing the card to the latest firmware resolved it.
I did the upgrade, I will make another test when appropriate.
I will also upgrade my curent card, I am curent at version 25, wich dates
2007, it might solve
I just wanted to add my voice to this attack. I saw the morning that I had
200+ distinct ips since the weekend. I used a small perl script that blocks
failed usernames and passwords at iptables level I found thei morning :
Here, we have 2 aastra 9133i and on 9480i ct cordless. We use often the park
call feature of asterisk to transfer calls to one another.
But the 9480i ct cordless cannot pickup a parked call. When manually
entering 701 (parked call extention), the phone display Call failed (appel
écoué in
Make sure your verbosity is set to at least 5 and try to see the CLI
output
on failure again. Are you sure the call is parked on 701 (not 702-720 as
defined in features.conf)?
Yes, after I can pick it up from my phone (9133i), and it works. I had
verbosity at 6 at the moment of testing.
Is the phone defined as a SIP extension/peer? If so, try sip set debug
peer xxx and try the call/pickup again.
Yes, and doing so, the phone could no longer dial out, bizare.
Yes, after I can pick it up from my phone (9133i), and it works. I had
verbosity at 6 at the moment of testing. When
Hi !
Sorry if this is a long post...
I had this setup for about a year without problems :
Network A - wrv200 - internet - wrv200 - net b
The 2 networks are linked with an ipsec vpn. The 2 internet connections are
with the same cable company to minimize latency, both separates /24
I personnlay found that marc is better than google when searching mailing
lists :
http://marc.info/?l=asterisk-usersr=1w=2
What is the best-recommended resource for searching archives of this
mailing
list?
Thanks for your time
___
-- Bandwidth
-- Executing [EMAIL PROTECTED]:4] NoOp(SIP/224-09e1d728,
Nicolas Ross 224) in new stack
-- Executing [EMAIL PROTECTED]:5] Wait(SIP/224-09e1d728, 0.5)
in new stack
-- Executing [EMAIL PROTECTED]:6] Dial(SIP/224-09e1d728,
SIP/225|15) in new stack
-- Called 225
-- SIP/225-09e73388
I cannot tell for sure for any system, but we have an old Portmaster PM3
hooked-up from one port of our Sangoma A104d card, another one being from
telco.
So, yes you can emulate the telco from a sangoma A10x card. Here's what I
have in my zapata.conf :
;Sangoma A104 port 1 [slot:12 bus:0
Hi !
We are running our asterisk from a transcend ts8gifd25. The whole system,
including the OS fit in this 8 gig disk. If you don't do any recording of
calls, you don't need that much of speed.
We have 20 or so SIP phones, a PRI trough a quad-port sangoma card, one other
port is a
I have a situation here where a user has an AAstra 480i phone, which
function corectly. The phone is behing a nat-router (a linksys wrv200 for
it's VPN point to point facility). The phone is plugued in a port wich has
qos enabled.
And when the user places a call, sometimes (not always), we get
Here, we've tested in the past zimsms, but now the,re closed. We founded out
that the most reliable and cost-effective way to send sms is with GPRS
modem. Multitech manufacures excelent quality product for gprs, edge and
others.
So we opened up an sms-only acount, it costs us 10$ / month for
the trick.
Thanks,
Nicolas
- Original Message -
From: Nicolas Ross [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, November 01, 2007 10:31 AM
Subject: Re: [asterisk-users] Connection astrisk to a RAS (portmaster
:57 -0400, Nicolas Ross wrote:
I also get sometime :
== Primary D-Channel on span 2 down
[Oct 31 20:50:53] WARNING[10250]: chan_zap.c:2393 pri_find_dchan: No
D-channels available! Using Primary channel 48 as D-channel anyway!
== Primary D-Channel on span 2 up
I don't claim
Here's my planed setup :
PRI from telco -- (port 1 of A104d) * (port 2 of A104d) -- PM3
The PM3, for those who don't know is lucent's portmaster RAS dial-up router.
I had setup asterisk, zaptel, libpri, wanpipe (as I have sangoma's cards).
In wancfg, I have port 1 as TDM_VOICE, with hardware
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