Hi,
No matching peer for '+331MYCLID' from '84.xx.xx.72:5060'
This line is telling you everything. The peer you've declared isn't being
matched for the incoming call and hence it tries to look in default
context (I assume allowguest=yes in your sip.conf)
Make sure that your peer is matched,
I see extra/additional fields in the pasted configuration
dahdi-channels.conf, try removing these. Then do a module reload
chan_dahdi
callerid=
group=
Regards,
Sammy
On Sun, Apr 22, 2012 at 12:49 PM, Satria Anamarta
anam.satri...@gmail.comwrote:
This is a very strange problem (at least
Thanks Sammy for looking at my post :)
Removing those 2 lines for all the channel still doesn't solve the problem :(
Any other ways I should try?
BR,
Anam
On 4/23/12, SamyGo govoi...@gmail.com wrote:
I see extra/additional fields in the pasted configuration
dahdi-channels.conf, try removing
do a complete reload(dahdi/asterisk/amportal) and then check.
On Mon, Apr 23, 2012 at 12:18 PM, Satria Anamarta
anam.satri...@gmail.comwrote:
Thanks Sammy for looking at my post :)
Removing those 2 lines for all the channel still doesn't solve the problem
:(
Any other ways I should try?
Restart the machine just now, and the caller id problem still there :(
Is there any other ways I can try again?
Thanks a lot :)
On 4/23/12, SamyGo govoi...@gmail.com wrote:
do a complete reload(dahdi/asterisk/amportal) and then check.
On Mon, Apr 23, 2012 at 12:18 PM, Satria Anamarta
On Sunday 22 April 2012, Satria Anamarta wrote:
This is a very strange problem (at least for me). I just realized that
started from April 20th 2012 every inbound call is from unknown.
Prior that, asterisk succesfully displayed the caller caller's ID for SOME
of the calls (30-50% success rate).
Thanks ricardo for your reply :)
I will try to comment that line. One thing for sure: the cid is
working on a analog phone, I just test it this morning.
Btw, somebody told that this may be caused by irq conflict. Is it safe
to disable the audio card on bios ?
Thanks again :)
On 4/23/12, Mc
On 22 Apr 2012, at 18:34, Steve Edwards wrote:
On Sun, 22 Apr 2012, Stuart Elvish - IP Exchange Systems wrote:
1. DO I need a separate server for the conference server?
This depends on a few factors:
(a) You won't be able to run MySQL alongside Asterisk with conferencing
and get good
On 04/21/2012 04:07 PM, bilal ghayyad wrote:
Dear;
The output of the ./configure that is related to dahdi is:
checking for DAHDI_RESET_COUNTERS in dahdi/user.h... yes
checking dahdi/tonezone.h usability... yes
checking dahdi/tonezone.h presence... yes
checking for dahdi/tonezone.h... yes
And
The Asterisk Development Team has announced security releases for Asterisk
1.6.2,
1.8, and 10. The available security releases are released as versions 1.6.2.24,
1.8.11.1, and 10.3.1.
These releases are available for immediate download at
Asterisk Project Security Advisory - AST-2012-004
Product Asterisk
Summary Asterisk Manager User Unauthorized Shell Access
Nature of Advisory Permission Escalation
Asterisk Project Security Advisory - AST-2012-005
Product Asterisk
Summary Heap Buffer Overflow in Skinny Channel Driver
Nature of Advisory Exploitable Heap Buffer Overflow
Asterisk Project Security Advisory - AST-2012-006
Product Asterisk
Summary Remote Crash Vulnerability in SIP Channel Driver
Nature of Advisory Remote Crash
Hi
I'm not agree problem could be cause from IRQ setting, I think in that way
problem should be more unstable, moreover no voice communication problem with
DAHDI service start up.
Best regards
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
--
Don't know about 1.8 but in 1.4 dahdi_genconf would update users.conf which
could mess with caller ID. Check your users.conf. If that's the problem,
you can fix and just do sip reload.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Are you able to add a Wait(2) at all to the beginning of your incoming
dialplan? A lot of missing callerID problems are because the callerID
value gets sent after the initial call signaling comes in.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com http://www.selbytech.com
--
Hi,
I hope for a hint on this issue.
I had a voicemail running on ast release 1.6.2 latest which i upgraded
to 1.8.11 latest release.
during this process I did add a couple of fields like minsecs and maxsecs.
I do now get empty emails where the attached voicefile only contains the
voice
If you are using Grandstream GXP 21XXX and 14XX phones and you are doing
any kind of remote firmware updates or config updates DO NOT use the
1.0.3.30 BETA version. We have found a bug in it that causes HTTP updates
to not work if you are using a domain name and not an IP address for
pulling
Thanks AJS :)
The different is only this:
21,22c21,22
rxgain=8.0
txgain=8.0
---
rxgain=0.0
txgain=0.0
29a30
Currently is rxgain and txgain set to 8.0 versus previous rxgain and txgain
= 0.0. I remember that because I change the txgain and rxgain setting the
CID is sometime appear (3~5 times
Hi Ricardo,
About the IRQ conflict and caller ID feature, I refer to this page:
http://www.stpaultechies.com/index.php?option=com_contenttask=viewid=83
from cat /proc/interrupts I notice that 2pcs of my Digium AEX800B (pci
express) card share a same IRQ
[root@callcenter asterisk]# cat
Thanks Danny :)
I dont see there is any wrong with users.conf
This is the contain of users.conf :
;
; User configuration
;
; Creating entries in users.conf is a shorthand for creating individual
; entries in each configuration file. Using users.conf is not intended to
; provide you with as much
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