I recently purchased a tigerjet usb phone with the TIGER560B chip($40). I
needed a portable phone that avoided my crappy laptop soundcard. Their
salesman said the phone is supported on linux and asterisk support would be
coming... When installing the linux drivers, the make crapped out. After
Hello,
There is an agi script for this, but I use goto's in the extensions.conf. Its
not terribly efficient, but it gets the job done.
I tried the blacklist but it only payed attention to the callerid. The number
was completely ignored. I could only put in one WIRELESS CALLER, and there
are a
oops. sent this last night - I was getting this list on an alias that I didn't
send from.
Hello,
I am attempting to setup spandsp on debian-unstable-2.6.7-recent-HEAD. Gnome
requires libtiff3g 3.6.1-1. I downloaded libtiff 3.5.7 and 3.6.0 and was
able to install 3.5.7,spandsp, and reinstall
Hello,
I receive an error during the make of spandsp on a sparc machine. I have
tried it with libtiff 3.5.7 and 3.6.0, but get the same error.
spandsp/mmx.h: In function `mm_support':
spandsp/mmx.h:72: error: unknown register name `edx' in `asm'
spandsp/mmx.h:72: error: unknown register name
cordless phone (out of 5) developed a bad key. I really have to
press hard to get the 1 key to work. The key went bad after 5 months.
Overall, I really like the phones.
Adrian Serafini
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Hi,
We use Opensips and like the results. The forks are similar, docs from
one can help in the other. The opensips mailing list is monitored by
one of the main developers. He is even in the IRC chat in the mornings.
The docs are kept current on the opensips webpage. They like to change
Is there a problem having 2 telcos on the same PRI card?
I think you go with one master timer as the Telco. Then the other spans
are secondary, tertiary, quaternary timers.
Adrian
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the monit daemon
to check for red alarms in syslog, then shutdown asterisk, then shutdown
the PRI, the backup PRI is auto switched through the Dataprobe.
Adrian Serafini
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AsteriskNOW is a GUI on top of Asterisk; it does not change the ability
of the system to handle call load.
I thought the AsteriskNOW GUI was now a FreePBX clone. If so, every
call now uses a perl script to make the call. This is considerably more
overhead than a dial-plan written in
Exactly, mixed audio, callers are linked to the call of another
caller,the calls are interlaced, is something that happens sometimes...
It can happen with analog dahdi calls. If this is the case, start
inbound on one end of the group, outbound from the other end.
Adrian
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On 09/13/2013 04:12 PM, jg wrote:
Is there a general recipe to avoid fraudulent calls under the
following conditions?
A receptionist transfers calls as a callee (customers are calling) and
as a caller (boss asks to call and then transfer to him), i.e. the
Dial cmd for the internal context
faster than using MySQL. Has anyone ever benchmarked this to quantify
Put Mysql on another machine and network the db service.
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New to Asterisk? Join
The qualify is on for the peer. It is failing to reply to the requested
SIP status. Maybe it is on wifi, screen goes off, wifi follows, zoiper
iax stack doesn't re-reg with the asterisk.
[Oct 8 18:14:14] NOTICE[510]: chan_iax2.c:11071 socket_process_helper:
Peer 'n4' is now REACHABLE! Time:
On 12/31/2013 12:41 PM, Vladimir Mikhelson wrote:
Nick,
You may want to try *97 and *98 to access voice mail.
Regards,
Vladimir
On 12/31/2013 10:23 AM, Nick Olsen wrote:
Greetings all, First time poster, Sorry if this has been answered here
before.
We recently replaced a failed 1.4x
On 01/13/2014 11:39 AM, Shaun Ruffell wrote:
If you have another board, yes, you could try. But I would recommend
checking all your cables, etc. Also, while highly unlikely, I've
heard of cases in the past where some smaller providers were
expecting to source timing from customer premise PBX
On 01/14/2014 04:32 AM, Olivier wrote:
I'm 100% sure my PBX is configured to use provider's clock (but I won't
swear my PBX is currently using provider's clock)
I have had to power the server down, UNPLUG the power, leave unplugged
for 4 minutes, power up. I had a T1 timing issue this
On 01/21/2014 01:55 PM, Stanley van Dijk wrote:
Hi,
Am running a freepbx install and created trunks, extensions and groups.
Now I'd like to hand out the Asterisk phonebook to the phones (all VVX
310's). Is there an easy way to do this?
Best,
Stanley
Even the old ones could view a webpage.
On 03/10/2014 07:39 PM, Thomas Rechberger wrote:
no trunking or bonding involved, so why just everybody calls this a trunk?
It is just another SIP peer. You tend to route more than one extension
down/from it.
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Im trying to modify the 'From' field in my sip headers in order to
include extra info (user=tel) as it follows:
The default extensions.conf has this, it might help.
;---
; from-pstn-to-did
;
; The context is designed
If h261 is checked in ekiga's video format list I have video, and
[Mar 21 16:25:32] WARNING[31818][C-0010]: file.c:1241
ast_writefile: No such format 'h261'
Ekiga can do SIP. Maybe try that? And set/prioritize the codec in
ekiga to desired codec, not h261.
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Coincidentally, 512 is my target. Any clues on how to get 200 more?
Upgrade to 1.4? hehe, I thought you were the self proclaimed 1.2
luddite? I'm a big fan of older releases with 1 year plus of uptime.
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On 03/21/2014 02:09 PM, David Woodfall wrote:
H.323 is a communications protocol like SIP. H261 is a codec like
ulaw or gsm. You do not need H323 unless you are using the H323
protocol INSTEAD of SIP.
I see. In Ekiga video codec window they are listed like:
[ ] h26190kHz H.323. SIP
to/from polycom and cell phone.
I didn't test: mms, a Q'ing mechanism if the cell/VoIP phone were
unavailable. I used the native sms app, it never failed on about 20 tests.
Adrian Serafini
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But the phone rings - so its routed - just no audio.
The ringing is SIP signaling. The audio is RTP data. See if the audio
is getting routed with a sniffer. Maybe use one codec that both clients
support.
Adrian Serafini
Maybe shut off qualify for the peer? I think I tried twinkle a few
years ago and it didna (yes didna) like the qualify packet. the sip
options qualify packet is only needed to keep the UDP state tables in a
firewall if the peer is remote
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On 05/24/2015 11:01 PM, Mehdi Shirazi wrote:
Hi
I want to load balance SIP calls between two(or more)
Asterisks with only DNS SRV. I used bidirectional sync
Unison to synchronize configuration files and internal database file
between two Asterisk boxes.
The problem is when a calls come to
it's mind
when codecs were changed and they did not exist in the config file.
Maybe the config file was changed during the upgrade?
Maybe test it with a fresh voicemail db?
Adrian Serafini
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