So I'm now using asterisk 1.8.5rc1 for Asterisk. I'm still getting
mysterious dropped calls. This only happens on calls that are outbound
on Dahdi and mostly happens in conference calls particularly
8xx-xxx-
This is the output of the hangup.
[Ksebpbx1*CLI
[0KPRI Span: 1 q931_hangup:
I interconnect the Asterisk and the Siemens PBX with Pri QSIG. But i
can't show the callerid name in the way Asterisk == Siemens. I
realized that Asterisk send calleridname in format
namePresentationAllowedSimple to Siemens e Siemens send calleridname
in format
.
Richard
2011/7/1 Richard Mudgett rmudg...@digium.com
I interconnect the Asterisk and the Siemens PBX with Pri QSIG. But i
can't show the callerid name in the way Asterisk == Siemens. I
realized that Asterisk send calleridname in format
namePresentationAllowedSimple to Siemens e
Hello, I have Asterisk 1.6 running on Centos, Also I have one analog
telephone line coming on
Wildcard TDM400P REV E/F Board 5
I can't get asterisk to dectect call coming from analog line.
Here is my /etc/dahdi/system.conf
fxsks=1
# global data
loadzone = us
defaultzone = us
[trunkgroups]
[channels]
switchtype=qsig
context = from-pstn
group = 0
signalling = pri_cpe
channel = 1-15,17-31,32-46,48-62,63-77,79-93,94-108,110-124
Everything after the channel line above will have no effect on the
channels created by the above line. Thus the faxdetect=both below
My setup is:
Asterisk1 w/ HA8+B400M --- Asterisk2 w/ HA8+B400M --- Patton SN4638
Asterisk1 is in TE/PtP (with termination)
Asterisk2 is in NT/PtP( with termination)
Patton is in TE/PtP
The cable between Asterisk boxes is an RJ11M-RJ11M (custom made with
pinouts 1 to 1, 2 to 2, ...).
The
The link between the Asterisk boxes must not be terminated on both
ends. Termination resistors on both ends is practically guaranteed
to cause link issues.
Very interesting but I'm afraid I still don't get it.
For various reasons, my goal is build an asterisk + patton solution
that
I have discovered that if I enable pickupsound = beep in
features.conf,
if I try to do a pickup with *8, the calling channel keeps on ringing,
while the phone where I pick-up from shows that the call has been
answered (I don't know where though). Also, it seems to completely
bugger up my
I am getting events using asterisk 1.4.41. However, when I place a
call
on hold I do not get that event.
Some of the events I am getting I show below. I wish to monitor when
channels are placed on hold
and taken off hold.
Set callevents=yes in sip.conf to enable Hold/Unhold AMI call events
Hi All
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged
isdn
channel in 1.6 and 1.4 using the following;-
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged
isdn
channel in 1.6 and 1.4 using the following;-
[snip]
Could be a problem in the media stream handling not being setup for
digital mode.
..., should I report a bug on
I'm using ISDN30 for a bridged application
in all the old versions of asterisk the time slot number is shown in
the channels and dstchannel fields of the cdr
I understand this has chaned in 1.8,is there a way of getting the time
slot information stored somewhere at the end of the call so
Hi All
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged isdn
channel in 1.6 and 1.4 using the following;-
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten =
Extracted from chan_dahdi.c:
Dial(DAHDI/pseudo[/extension[/options]])
Dial(DAHDI/channel#[c|rcadance#|d][/extension[/options]])
Dial(DAHDI/subdir!channel#[c|rcadance#|d][/extension[/options]])
Dial(DAHDI/ispan[/extension[/options]])
Is it possible that the installation libpri-1.4.11.5 newer than the
libpri-1.4.11.5-patch?
Well, when I typed (note: I am trying to apply the
libpri-1.4.11.5-patch for the libpri-1.4.11.5):
libpri-1.4.11.5# patch -p0 -i libpri-1.4.11.5-patch
It gave me that patched detected as shown
But believe me no one option works for me. I tried dahdi/25/XXX
but it still using pri first channel or anyother channel
Dialing DAHDI/25/ will dial channel 25 only.
For ISDN spans I do not recommend doing that because your call will
fail if that channel is already in use when there
We have two pri line and I want to see how asterisk distribute
outgoing call per channels
I meant it use first last channel 47 or it will use first channel?
Or it will allocate dynamically ?
Extracted from chan_dahdi.c:
Dial(DAHDI/pseudo[/extension[/options]])
As I'm reading this, libpri thinks that the SV8300 is complaining that
a mandatory IE is missing, in this case time/date. However, the
field is
THERE. But when I go back to a working libpri (r1878), I see that the
time/date is NOT sent on the CONNECT.
If I'm reading Q.931 correctly, 5.1.8
Please create a mantis issue describing this problem.
Pardon my ignorance, but what does mantis refer to?
Mantis is the issue tracker at:
https://issues.asterisk.org
Richard
--
_
-- Bandwidth and Colocation Provided by
301 - 319 of 319 matches
Mail list logo
