Callwithus.com is discontinuing iax service.
Can anybody recommend IAX provider - I need somebody with good rates to
Philippines.
--
Joseph
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Hello all,
how can I possibly make the monitoring for all calls through the
asterisk, and for those file to be stored with the name of the initiator, in
additional to know to whom this call is going, could this functionality be
implemented via configurations!
in other words, could I
Hi we have set up two asterisk machines where we do all the managing of SIP
calls. Now, sometimes we call and we get an underlying sound that is a
locution from a customer. What could make this to happen ? Is very strange
to us, but maybe we are missing something... some configuration, or anything
Sorry for dropping in so late, but maybe our solution do configure snom
phones can help you
We have written a small script that scans the network for snom phones.
This is done by doing broadcast pings and using the arp-scan command and
reading the arp cache. Then we filter the results on the mac
Alex Samad wrote:
Voltage isn't the issue - the difference is in the impedance. Australia
I get this in my dmesg when I load up the rdm410 modules
[1083334.103487] Freed a Wildcard
[1083336.171371] ALAW override parameter detected. Device will be
operating in ALAW
[1083338.040522]
Hi, we are experiencing a problem that is very strange, only on SOME calls,
a locution jumps in to the RTP stream and both persons between the phones
can hear it. It is looped and it does not stop till hang up. Do you have any
clue about what could be happening ?
Thank you !
Trixbox I think uses FreePBX
FreePbx has an option for each extension to set it to record all calls.
It will record the extension in the file name and you can view it
through the recordings app if you want a web view.
There are all stored in a common dir /var/spool/asterisk/monitor - you
can
Please post ONCE to the list.
Please define the 'locution'. What words can you hear? (BTW locution
is a rather uncommonly used word in english).
Steve
On 29 Jun 2009, at 10:07, Xavier Cardil wrote:
Hi, we are experiencing a problem that is very strange, only on SOME
calls, a locution
Hi, do somebody knows how to sniff RTP and SIP traffic only for a faster
debugging ?
Thanks.
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I meant that we hear an audio stream typical from an IVR application, that
says press 1 if you want to . . blabla press two if you want to . . . We
have checked the configuration and the code of our IVR application but we
can't see why this is playing, and only in some calls, not in all calls. We
For Linux use tcpdump on the host you are after
tcpdump udp and port 5060 or portrange 1-16000 -s0 -i eth0
where 5060 is your SIP port and 1-16000 are your rtp ranges
-s0 means snap length of 0 so capture all the packet rather than cutting
off at a point
And refine it by adding the
Thank you so much !
On Mon, Jun 29, 2009 at 12:21 PM, Duncan Turnbull dun...@e-simple.co.nzwrote:
For Linux use tcpdump on the host you are after
tcpdump udp and port 5060 or portrange 1-16000 -s0 -i eth0
where 5060 is your SIP port and 1-16000 are your rtp ranges
-s0 means
On 29 Jun 2009, at 11:00, Xavier Cardil wrote:
I meant that we hear an audio stream typical from an IVR
application, that says press 1 if you want to . . blabla press two
if you want to . . . We have checked the configuration and the code
of our IVR application but we can't see why
No, it doesn't sound like any of our IVR applications. We have 2 two Cisco
AS5400 PSTN to SIP gateways, so yes we do outbound / inbound calls.
Thank you.
On Mon, Jun 29, 2009 at 12:52 PM, Steve Howes st...@geekinter.net wrote:
On 29 Jun 2009, at 11:00, Xavier Cardil wrote:
I meant that we
My guess would be the fault is on the analogue side not the SIP.
Steve
On 29 Jun 2009, at 12:05, Xavier Cardil wrote:
No, it doesn't sound like any of our IVR applications. We have 2 two
Cisco AS5400 PSTN to SIP gateways, so yes we do outbound / inbound
calls.
Thank you.
On Mon, Jun
I received a phone call asking for specs, how I'd use, etc, etc and they
said they'd be turning my beta account up in another 6 weeks.
That was 3 weeks ago.
PB
On Mon, Jun 29, 2009 at 1:31 AM, randulo spamsucks2...@gmail.com wrote:
Though they have written me back twice to say coming soon I
Hi all!
My problem is that calls being placed in the queue, and are waiting
while the agents are busy, when an
agents is then free they gets connected to the agent but there is
silence (no voice).
If a caller has not to wait in the queue, there is no problem.
My agents have an iax2 client, and
When doing transfers the call drops as follows:
1. I receive a call (internal or not)
2. I dial *2, wait for transfer sound plus dialtone
3. I dial for destinantion person, who pickups the phone
4. We talk to each other
5. I hangup my phone and the call drops
if I dial * when talking with
Hi everyone,
This is my first post, so apologies if I have not included all details
about the issue. I am using a Nokia e71 to connect to a corporate
asterisk server and am having issue with dialing. I can dial all
extensions and receive all types of incoming calls. I cannot however,
dial
Sorry i replied late bcz i have to do some other work
I have a new required functionality. that is
Develop a Client server application that will communicate using a
normal modem with out connecting to internet.(Client with a PC and
modem will dail the number of server it will be a PSTN number (Not
You have tried putting # after the number (for example 5551212#)? You could
have a dialplan problem.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kayton Sapale
Sent: Monday, June 29, 2009 8:49 AM
To:
I have used an nokia n95 with asterisk without any problems (except for the
actual phone deciding to restart itself every few hours - but that's nothing
new with nokias!)
Are you getting anything on the CLI that might point you in the right
direction when the call is attempted?
CHeers
2009/6/29
Hi all,
i would like to ask please about how to force asterisk to ask for
authentication when receiving an INVITE packet from any device?
Regards
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That's the strange thing. Nothing shows when monitoring the service in
debug. On the phone, however, I do see a connection time-out error.
I guess this might indicate that the device is attempting to connect to
the service in a way different from when just dialing an extension?
Geraint
Hellow,
* I have a problem with dial up signalling. currently I have configured
asterisk server and E1 card to ISP. then other side I am having ATA to PC
for connecting internet through DialUP connection. is it possible and please
send me the procedure how I can do it ?? *
ISP - Asterisk - ATA -
IMO, it is indeed connecting differently. When you dial an extension, say
1000, you get a match connection. When you dial the local number, you go
to a different segment of the dialplan. You say that other softphones
connect to local numbers correctly; have you checked things like truncation,
This is the configured pattern for local calls - _NXXNXX
When I dial a local number from the device, I dial a number like
7706743900 and select internet call. Does what I dial not match the
pattern?
Danny Nicholas wrote:
IMO, it is indeed connecting differently. When you dial an
Your pattern appears to be set up in anticipation of a leading digit. What
happens if you dial either 17706743900 or 97706743900?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kayton Sapale
Sent: Monday, June 29, 2009
It does this by default unless you have allowguest set to yes, and/or
any insecure parameter options on any individual peers.
--
Sent from mobile device
On Jun 29, 2009, at 10:33 AM, michel freiha mich...@gmail.com wrote:
Hi all,
i would like to ask please about how to force asterisk to
I tried the following:
17706743900
97706743900
On both, I receive the same timeout message. I also tried with
different numbers as well, just to give a try, same results. Just for
giggles, I tried:
_7706743900
When trying this, I get a Address not in use message, which I think
means
Without getting into a lot of detail, this will not work. Period.
You just can't do reliable modem passthrough with VoIP in most cases,
some clever proprietary hacks notwithstanding.
To the extent it is possible, nobody is going to send you the
procedure.. This list is for specific
One to few X's for that number?
Cary Fitch
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Monday, June 29, 2009 10:10 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re:
After reading the Asterisk PDF again, your pattern is fine, it's probably
just not in the right place in your dialplan. Check your sip.conf and
users.conf to insure that the device hits the correct context to dial out.
You might want to post them here for a quick checkup.
_
From:
jonas kellens wrote:
asterisk*CLI sip show domains
Our local SIP domains: Context Set
by
jocan.local (default)
[Configured]
192.168.1. (default)
[Configured]
James Lamanna wrote:
I remember seeing a T38 Gateway application for Asterisk 1.6 floating
around, but I can't seem to find it again.
Does anyone have any pointers to it? I really want to be able to send
an incoming T38 connection directly to the PSTN.
You might be looking for this issue in
Looking for a (windows) app. that will listen to the manager interface
then pop-up a web browser pointing to a page on an incoming phone call..
Not looking for outlook integration, or outbound dialling, just to
recognise an incoming call and poke a URL at a website in a browser and
I've
On Mon, Jun 29, 2009 at 2:49 AM, Joseph syscon...@gmail.com wrote:
Callwithus.com is discontinuing iax service.
Can anybody recommend IAX provider - I need somebody with good rates to
Philippines.
--
Joseph
___
-- Bandwidth and Colocation
On Mon, Jun 29, 2009 at 2:49 AM, Joseph syscon...@gmail.com wrote:
Callwithus.com is discontinuing iax service.
Can anybody recommend IAX provider - I need somebody with good rates to
Philippines.
--
Joseph
Did you ask why they are dropping IAX2? That is a fairly big decision.
I assume
jonas kellens wrote:
Problem 1 :
Incoming conversations from the SIP-provider come into the
[default]-context and to the 's'-extension.
I am unable to change this, even if I have :
I have the same (or similar) issue with one of my ITSPs.
In my case, the problem seemed to be because they do
Not related to asterisk: but I figured someone here would have used them
before?
Has anyone tried http://www.wcell.com/ab/ZnJpZW5kLzEwNjI= yet?
Looks like they have a voip app for your mobile handset sending voice
calls out over your data service or wifi for 1c per minute calls both
URL reeks of affiliate link.
http://www.wcell.com/index.php gets you to the same place your
/ab/ZnJpZW5kLzEwNjI http://www.wcell.com/ab/ZnJpZW5kLzEwNjI= affiliate
link does.
Not sure how you navigate to get the URL you posted.
Affiliate = commercial
Thanks,
Steve Totaro
On Mon, Jun 29, 2009
http://www.wcell.com/incentives.php
MLM, uuggg
I have some Amway stuff you can buy too!
On Mon, Jun 29, 2009 at 2:26 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
URL reeks of affiliate link.
http://www.wcell.com/index.php gets you to the same place your
On Mon, 29 Jun 2009, Gordon Henderson wrote:
Looking for a (windows) app. that will listen to the manager interface
then pop-up a web browser pointing to a page on an incoming phone call..
Not looking for outlook integration, or outbound dialling, just to
recognise an incoming call and poke
2009/6/29 Loic Didelot ldide...@mixvoip.com
Sorry for dropping in so late, but maybe our solution do configure snom
phones can help you
We have written a small script that scans the network for snom phones.
This is done by doing broadcast pings and using the arp-scan command and
reading the
Just found out from Mitchel Constantin via a Facebook reply that his
company installed the Freeswitch backend backend for Wcell.
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
mailto:d...@cognation.net +1-212-203-4357 New York
+61-2-9016-5642 (Sydney in-dial).
I was trying to enable CDR in a mysql-database when the following
occured :
asterisk*CLI cdr status
CDR logging: enabled
CDR mode: simple
CDR output unanswered calls: yes
CDR registered backend: cdr_manager
CDR registered backend: cdr-custom
asterisk*CLI exit
[Jun 29 21:56:52] Executing last
On Thu, Jun 4, 2009 at 11:08 AM, James Lamannajlama...@gmail.com wrote:
Hi,
I have a serious problem with Asterisk 1.4.18.
Every so often, usually after Asterisk has been running for a few days
consistently, phones start dropping registrations.
However, when this happens, doing a sip show
Why can't you just do a daily/weekly cron to restart when convenient in
off/slow hours for local time. Is your business constantly on-line 24/7?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Lamanna
On Mon, Jun 29, 2009 at 10:22:54AM +0200, Loic Didelot wrote:
Sorry for dropping in so late, but maybe our solution do configure snom
phones can help you
We have written a small script that scans the network for snom phones.
This is done by doing broadcast pings and using the arp-scan
On Mon, Jun 29, 2009 at 4:23 PM, James Lamanna jlama...@gmail.com wrote:
On Thu, Jun 4, 2009 at 11:08 AM, James Lamannajlama...@gmail.com wrote:
Hi,
I have a serious problem with Asterisk 1.4.18.
Every so often, usually after Asterisk has been running for a few days
consistently, phones
Xavier Cardil wrote:
Hi, do somebody knows how to sniff RTP and SIP traffic only for a faster
debugging ?
I'm not sure what you mean by for a faster debugging. As for sniffing
the traffic, tcpdump works well.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel:
Hello,
I think you do not understand. The phone gets an IP from the DHCP
server. Nothing special in this step.
Then you save the configuration url to the snom by calling the url I
showed you in my previous e-mail. This can be done using: wget, lynx,
links, php, perl ..
Once the phone has the
BellSouth (now ATT) has a number you can dial and it will play back
voice prompts with your calling number? It's used by their techs with a
buttset in identifying analog 1FB lines...
Eg Dial 704-210-3233, it answers
Seven-Zero-Four-Five-Five-Nine-Two-One-Two-Two if your dialing from
On 06/29/09 13:54, Steve Totaro wrote:
On Mon, Jun 29, 2009 at 2:49 AM, Joseph syscon...@gmail.com wrote:
Did you ask why they are dropping IAX2? That is a fairly big decision.
I assume it was eating more resources than it was worth.
Sort of like IAX.cc (Vitelity) recommending IAX2.
I have
To clarify the question is what is the number for ATT Calling Number
Verification?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason
Aarons (US)
Sent: Monday, June 29, 2009 7:24 PM
To: Asterisk Users Mailing List - Non-Commercial
Why can't you just do a daily/weekly cron to restart when convenient in
off/slow hours for local time. Is your business constantly on-line 24/7?
I have tried that. Unfortunately restart when convenient doesn't
always seem to actually restart
asterisk, presumably because there are stuck calls
Hi All
I am using asterisk 1.4.21.1
Im not sure if this is a issue but it has become one for me :)
When agents are logged in to a queue (AgentCallBackLogin) and they receive a
direct line call or a transfer they still receive queue calls.
EG
Someone in our company transfers a call to a
On Mon, Jun 29, 2009 at 8:13 PM, Josephsyscon...@gmail.com wrote:
Is SIP really so much better than IAX2 that providers are dropping it?
The major issue for me with SIP is firewall traversal.
I wouldn't say 'better' so much as 'understood'. When I've worked with
SIP, including integrating
provided by mci
800.444.
Thanks,
Steve Totaro
On Mon, Jun 29, 2009 at 8:09 PM, Jason Aarons (US)
jason.aar...@us.didata.com wrote:
To clarify the question is what is the number for ATT Calling Number
Verification?
*From:* asterisk-users-boun...@lists.digium.com [mailto:
The queue option
ringinuse = no
might be what you are looking for.
PaulH
Kev Szaszvari wrote:
Hi All
I am using asterisk 1.4.21.1
Im not sure if this is a issue but it has become one for me :)
When agents are logged in to a queue (AgentCallBackLogin) and they receive a
direct line
Hello, all. We have successfully connected our new Asterisk 1.6.1.1 PBX
to Vitelity's network and have been very happy with them thus far.
However, we'd like to use domains in our sip.conf to facilitate routing
in our multi-tenant environment. We also like to set
allowexternaldomains=no for
It appears that that option is set
from queues.conf
[ops]
musicclass = default
strategy = leastrecent
timeout = 5
retry = 1
wrapuptime= 3
autofill = yes
autopause = no
maxlen = 0
joinempty = yes
leavewhenempty = no
ringinuse = no
- Original Message -
From: Paul Hales
Hello, all. I see that I can use the C flag in Dial() to reset the CDRs
and plan to use this for outbound calling since our carrier does not
bill us until the call is answered.
We'd like to do this for inbound calls from the PSTN since we pay for
inbound and outbound minutes. However, it
The strange thing is, Queue calls are working as per expected. If they get a
call from the queue they wont get another until the 1st call is done.
Its only when the agent received a direct call or a internal call from another
staff member, the queue continues to ring their phone.
-
Hellow,
I have cisco 7911 and 7906 worked with asterisk server. But i can not set
the time and date for these phones. can any one tell me how can i set the
time and date for these phone.
Thanks
mahboob
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Not true...
You can provided you disable data compression (ATK0) on your modem.
Reason? Because a codec is already compressed. Adding compression at
the modem level to an already compressed bitstream == lost bits. I call
all over the world all the time using asterisk/sip/ulaw with decent
I think the handling of this may have improved in later versions of
Asterisk - is an upgrade an option?
(I tested this with a newer version of Asterisk recently, and it behaved
how you were hoping it would behave)
PaulH
Kev Szaszvari wrote:
The strange thing is, Queue calls are working as per
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