Displaying 20 results from an estimated 900 matches similar to: "Calls stuck in the queue even when ext's are available"
2009 Jul 20
0
No subject
device somewhere in your communication path, and since voice is picked up as
DTMF, some device is also set to listen for inband DTMF.
What is the origination source of incoming calls to your system?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-07-08 4:24 PM, "das sandesh" <sandesh440 at gmail.com> wrote:
Hi,
We have few systems with asterisk 1.4.22.1 and we use sip
2009 Dec 14
3
Question regarding digital card TE412p
Hi,
I was able to implement T122p one port PRI and was able to call out, but I
am planning to use TE412p (includes echo cancellation) 4 port digital card
(PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI
connections) with proper hardware like dual core quadcore processor and 8gb
RAM in one server?
Also I was planning to implement using 64 bit architecture with Asterisk:
2009 Oct 21
4
Concurrent calls including mysql taking lot of time for execution
Hi,
I tried getting our server setup for 400-500 simultaneous calls, calls were
going through properly but at around 200-250 calls, mysql (connect ...)
statement was taking at least 5-10 sec to connect to the database. I
optimized all possible parameters in my.cnf:
max_connection=1000
wait_timeout=60
query_cache_type=1
query_cache_limit=4M
query_cache_size=512M
interactive_timeout=120
2010 Jul 08
2
DTMF issues/redial tones with rfc2833
Hi,
We have few systems with asterisk 1.4.22.1 and we use sip trunking for them
not PRI's, one of our system is giving a problem of dtmf (rfc2833), like
when we dial the number that have IVR and enter the extension or access
code, it some time takes it and some times does'nt recognize the digits
dialled. We also tried auto and info for dtmf but could not get the dtmf to
work reliably, can
2010 Mar 12
0
Regarding - P-Asserted identity and Privacy - SOLVED
Hi All,
I got this figured out, when the privacy is ON at the other end of the
server and when we get the Invite message to the server connected to PRI's,
just take the details from the invite message in the Dial plan and send the
calls as anonymous:
exten => _1NXXXXXXXXX,n,Set(PRIVACY=${SIP_HEADER(Privacy)})
exten => _1NXXXXXXXXX,n,ExecIf($["${PRIVACY}" =
2009 Oct 15
3
DS3 capacity calls using asterisk
Hi All,
We are trying to implement a DS3 capacity calls (672 concurrent calls) using
asterisk server. I wanted to ask are there any compatible DS3 cards with
asterisk? I tried searching a lot but could find DS3000P from digium but
unable to get this product. Does anybody have any idea of having any DS3
card in asterisk box so as to handle around 600 calls?
Thanks
Sandesh
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2010 Jun 11
2
asterisk log problem
Hi All,
We have built an asterisk server (asterisk - 1.4.26.2) where there would be
around 322 concurrent calls going on, but I can see that full log grows
rapidly, in one day it reaches to around 10-15 GB if I turn on the sip debug
and its tedious even by using any commands to get the required call from the
log if there is any problem. Is there any way of splitting the full log into
parts
2010 Jul 21
1
Redial dtmf tones randomly...asterisk 1.4.21.2
Hi,
We are experiencing this issue of redial dtmf tones generated randomly in
the Voip calls, we have asterisk 1.4.21.2, dahdi 2.0.2.2 and we have dtmf as
rfc 2833, we have a cisco router at the location Cisco 2431, 8FXS (only one
FXS is used for Fax and rest are empy) connected to the netgear switch and
all the phones are connected to this switch and there are no non sip devices
in the
2010 Feb 19
0
asterisk sudden restart - 1.4.18.1 - Resolved after upgrade to 1.4.22.1
Hi Leif,
Thanks for the information. I checked the /tmp/ folder and there was core
#### files and I tried to back trace it but it was not showing the cause of
that crash, but anyhow, I upgraded our Asterisk system to 1.4.22.1 and from
past few days its going on fine. I have also researched and found that
version 1.4.17/18.1 had the issue of channel stuck up as well as random
asterisk crashes.
2010 Jun 25
2
Call drops on group paging asterisk - 1.4.22.1
Hi All,
We are using group paging and our asterisk version: 1.4.22.1, but when ever
any one page to the whole group (28 extensions), the calls which are on hold
on those extensions will be dropped, is there any other way to have this
feature or to go with Overhead paging. Currently this has become a serious
problem, can anyone through some light on this group paging senario?
Thank you very much
2010 Oct 20
1
Parked calls drop asterisk-1.4.22.1
Hi
We are facing a problem for orphaned parked calls, we have the following
config:
asterisk -1.4.22.1
dahdi-linux-complete-2.2.0.2+2.2.0
and when we get an incoming call and after it gets parked, after some set
time (here its 2 min), it goes back to the operator, but the problem is that
randomly it tries to call SIP/5060 instead of SIP/2200 (where 2200 is the
extension number of the operator)
2009 Sep 19
1
"Channels got stuck in asterisk 1.4.18.1"
Hi All,
Today I faced a problem with channels getting stuck. We use asterisk
1.4.18.1, and there were 2 extensions (channels) that got stuck. When I try
to do "soft hangup <channel>", it says "Requested for soft hangup" for that
channel, but if we go and check once again those channels are still stuck.
Also even after asterisk restart it did'nt go, finally we had to
2008 Jul 02
3
Unable to switch input to xen from serial console
Hi all,
When i do ''xm dmesg'' the last statement says "*** Serial input ->
DOM0 (type ''CTRL-a'' three times to switch input to Xen)" (i have no clue
what''s that supposed to mean??) But when i press ctrl-a three times at
the serial console, nothing happens.
Iam using minicom to connect to the serial port of xen machine. Once
xen
2010 Jan 22
0
Handling SIP error codes/ISDN codes
Hi,
I was trying to use 2 of asterisk servers and interconnected, one of them as
a peer to other sever (configured in sip.conf), so all the calls to server 1
will just be passed to server 2 (has PRI Card, TE 412P, only one PRI
connected), i was sending calls to server 1 and that would send to server 2
and then dial out using Dahdi, but the problem that i got was the hangup
cause codes, i was not
2008 Jul 24
1
doubt on phys_to_machine_mapping
Hi all,
Can some one tell me where phys_to_machine_mapping is being
initialized for a domU having paging mode set to PG_translate.
I see that, populate_physmap() after calling __alloc_xen_heap_pages only
updates the machine_to_physmap but how is the mfn for the allocated page
being updated/set for phys_to_machine_mapping??
I see that phys_to_machine_mapping is a #defined to RO_MPT_VIRT_START
2009 Sep 25
1
"multiple contexts for multiple locations"
Hi All,
I have a senario where we have multiple locations and all have the ability
to call using 1NXXXXXXXXX pattern, so we have created multiple contexts so
the outbound goes fine, but while transfer occurs (after picking the inbound
call and transfer), it uses the first 1Nxxxxxxxxx priority patterned
context, like if the 3rd location is making a transfer, but 1st location
have the priority
2008 Aug 21
2
doubt on releasing domain pages
Hi,
I am trying to release domU pages from page_list and xenpage_list
after domU shutdown while retaining the rest of the domain information.
To achieve this in __domain_finalise_shutdown i call
domain_relinquish_resources. This is failing to release pages from
page_list for type PGT_l2_page_tables and crashing dom0.
To be specific, while testing on mini-os i saw that when
2010 Feb 10
1
asterisk sudden restart - 1.4.18.1
Hi,
Asterisk got stopped this morning after 20 minutes and phones went to 'No
Service' and then got started automatically after 20 min, as I could see in
the full log that asterisk got started at so and so time:
[Feb 10 08:29:31] VERBOSE[31013] logger.c: Asterisk Event Logger Started
/var/log/asterisk/event_log
[Feb 10 08:29:31] VERBOSE[31013] logger.c: Asterisk Dynamic Loader Starting:
2010 Mar 19
4
Call Drops while doing assisted transfer from remote location
Hi all,
We have our system hosted publicly and 4 phones are connected remotely at
employee's home, and when they try to do a assisted transfer to one of the
employee at the main office, the call is lost. For ex: person A calls person
B, person B calls person C for assisted transfer, and as soon as person B
hits transfer button again to transfer person A to C, the call is lost.
But in the
2009 Nov 05
0
SIP 503 instead of SIP 480 in asterisk debug mode
Hi All,
I was actually trying to use the dialplan application that uses 'Dial' and
when the: Dial(SIP/XXXXXXXXXX at xxxx|20|) command is executed and the
destination number rings for 20 sec after which I receive as "503 Service
Unavailable", but not "480 Temporarily unavailable".
Dial(SIP/XXXXXXXXXX at xxxx|20|)
exten => XXXXXX,n,NoOp(Dialstatus:${DIALSTATUS})