Michelle Dupuis wrote:
We're looking at a large wifi phone deployment, and we're looking for
wifi phones that:
1. Are SIP compliant (Asterisk friendly)
2. Provision capable (ideally TFTP of own MAC address)
3. Industrial quality (no cheap plastic stuff).
4. Well documented (and none of
Hi all,
I am working on
asterisk 1.2.18
zaptel 1.2.17
Polycom 650
polycom 430
SIP version 2.0.3.0131 for IP 650
SIP version for IP430 2.0.3.0127
freepbx 2.2.1
I am trying to configure BLF using asterisk but failed. I would be thankfull if
somebody help me.
Regards
FArooq
Hi all,
As we know we can configure in astersik like before 5:00pm calls go to
reception and after 5:00
pm calls go to some mobile no. One of my client requested that he wants to
manually shift the dial
plan like above as he has flexiable timing sometime he finishes at 3:00pm some
time 8pm.
Stephen Bosch wrote:
Russell Bryant wrote:
Lacy Moore - Aspendora wrote:
On 6/29/07, Ade Vickers wrote:
What I'd like to do is have the music streaming
constantly, so the on hold
caller always gets music at the current position; even if
that's in
the middle or near the end of a
Hi Russell
2007/7/2, Russell Bryant [EMAIL PROTECTED]:
I guess the right thing to do would be to try to contact the developer
directly.
Do you happen to know who that might be? You could answer using a PM.
Best regards,
Bent
___
--Bandwidth and
Hi Grigoriy,
If I'm interpreting your call flow correctly, you're only ever making one
outbound call - that's the call from ext100. You're then transferring that
call to ext200. Is this correct?
If so, given that you're only making one outbound call, then CDR is acting
as expected. You only ever
On Jul 2, 2007, at 11:56 PM, Ron Stephan wrote:
Please tell me how you can construe making a call with the the
CID of a number in your control to be Misleading or inaccurate
Sure - it goes like this - The less scrupulous among us might use a
spoofed cid to get people to do something
Andrew Joakimsen wrote:
The Proposed bill S704 reads It shall be unlawful for any person
within the United States, in connection with any telecommunications
service or IP-enabled voice service, to cause any caller
identification service to transmit misleading or inaccurate caller
Hi all,
As we know we can configure in astersik like before 5:00pm calls go to
reception and after 5:00
pm calls go to some mobile no. One of my client requested that he wants to
manually shift the dial
plan like above as he has flexiable timing sometime he finishes at 3:00pm some
time 8pm.
Hi all,
I am working on
asterisk 1.2.18
zaptel 1.2.17
Polycom 650
polycom 430
SIP version 2.0.3.0131 for IP 650
SIP version for IP430 2.0.3.0127
freepbx 2.2.1
I am trying to configure BLF using asterisk but failed. I would be thankfull if
somebody help me.
Regards
FArooq
On 7/3/07, J. Oquendo [EMAIL PROTECTED] wrote:
You're answering your own question. Forwarding a call with a number
that is not the originating number is what (drum roll)
And in a corporate environment, what is the originating number? Is it
the main line, the DID, or what?
If I am at my house,
Ooops did Google just become a carrier :)
http://googleblog.blogspot.com/2007/07/all-aboard.html
I hear stocks crumbling worldwide as I type.
Cheers,
Dean
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users
Lacy Moore - Aspendora wrote:
This all gets complicated, and there is not a US Representative or US
Senator smart enough to figure this out, that's the scary part. Most
probably don't even know what a DID is. By the time it is over with,
laws will be passed to outlaw legitimate purposes.
On 7/3/07, Farooq Ahmed [EMAIL PROTECTED] wrote:
Hi all,
As we know we can configure in astersik like before 5:00pm calls go to
reception and after 5:00
pm calls go to some mobile no. One of my client requested that he wants to
manually shift the dial
plan like above as he has flexiable
Asterisk 1.4.5 full log:
[Jul 2 09:31:16] VERBOSE[2682] logger.c: == Refreshing DNS lookups.
[Jul 2 09:31:16] NOTICE[2682] dnsmgr.c: host 'outbound1.vitelity.net'
changed from 64.2.142.17 to 64.2.142.29
[Jul 2 09:31:23] DEBUG[2711] jitterbuf.c: Attempting to exceed
Jitterbuf max 600 timeslots
Actually, I *NEED* to change the caller ID. Here's why...
Someone dials into my DID, their caller ID reflects their (cell, home
office, etc...)
The call then rings my VoIP phones.
It then announces Outside Transfer after 3 rings, at which time it
rings my VoIP phones AND my cell phone.
If PSTN
Many times the news does not carry information about bills before
congress. So the only time we hear about them is after the fact. I blame
the news in the US as they are the ones initiating the stupid Paris
Hilton stories instead of the Real news.
Bob R
J. Oquendo wrote:
Lacy Moore -
problem - occasional garbled calls, mostly remote users.
T1 connected to PSTN, SIP over local LAN and internet to remote users. NAT
at local firewall and at remotes. There is no traffic shaping in place, no QoS.
Most are Polycom phones, two Aastra's.
Start with QoS on LAN switches? No
On 7/3/07, J. Oquendo [EMAIL PROTECTED] wrote:
Reposted to this list: (http://lists.virus.org/voipsec-0610/msg00046.html)
That's exactly the type of thing that needs to be stopped. If Dell
outsourcing calls me from India, the CLI must be their number in India
not a faked-in number of some
On 7/3/07, Karl J. Vesterling [EMAIL PROTECTED] wrote:
And frankly, *NO*... I don't want to give anyone my cell number. Once
you give out the cell number, people call you on it before they attempt any
other number.
You are absolutely correct. I walk down the hall of our office and see
I have a cisco 7905 running sip code that is successfully connecting to my
asterisk system. I also have a softphone that is connecting to the system.
I can make a call from the cisco extension and the softphone rings.
However; I can not make a call from the softphone to the cisco extension.
it
Ade Vickers wrote:
*snipped
Hi all, thanks for the responses so far.
I too understood it to be a configuration thing, with the addition of a
streaming music server (which, obviously, provides the MoH stream). Asterisk
should then simply pick up the stream play it whenever MoH is
Dear list,
following problem, i have some users, who are supressing their callerid.
This setting is adjusted at the sip phone. So if these guys are calling
internal persons nobody sees the callerid. I am looking for the
following resolution:
User has set his phone to anonymous, user calls
Karl J. Vesterling wrote:
Actually, I *NEED* to change the caller ID. Here's why...
CID internal and external are two different things.
If PSTN gateway providers lock the callerid to my DID and I have no
way to change it, then I have no idea whom is calling me. And that is
a
Hi Olivier,
I forgot to mention it is a C450IP.
But if you have some hint on S maybe it can help me. Perhaps it is some
configuration...I tried with qulify=no as I read on a web page without
success.
Thank you.
Giorgio Incantalupo
Olivier wrote:
Is it a S 450IP ou C 450IP ?
Richard Lyman wrote:
Ade Vickers wrote:
*snipped
Hi all, thanks for the responses so far.
I too understood it to be a configuration thing, with the addition of a
streaming music server (which, obviously, provides the MoH stream). Asterisk
should then simply pick up the
hdpml wrote:
Dear list,
following problem, i have some users, who are supressing their callerid.
This setting is adjusted at the sip phone. So if these guys are calling
internal persons nobody sees the callerid. I am looking for the
following resolution:
User has set his phone to anonymous,
Hi Richard,
Thanks for those replies - I'll give them a shot shortly.
Cheers,
Ade.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Richard Lyman
Sent: 03 July 2007 16:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
These are the things you should check first:
1.) Make sure that your cable/line is not faulty.
2.) Make sure you are running the latest version of zaptel for your
particular branch (1.2 or 1.4)
3.) Make sure that your timing is correct for the span in zaptel.conf
Example:
(If it's a span from
On Jul 2, 2007, at 6:02 PM, Tzafrir Cohen wrote:
On Tue, Jul 03, 2007 at 12:04:08AM +0200, Administrator TOOTAI wrote:
Tzafrir Cohen wrote:
On Mon, Jul 02, 2007 at 09:27:39PM +0200, Administrator TOOTAI
wrote:
We have an Ubuntu Dapper with 2.6.14 kernel, asterisk 1.2.14
debs from
://lists.digium.com/mailman/listinfo/asterisk-users
__ NOD32 2374 (20070703) Information __
This message was checked by NOD32 antivirus system.
http://www.eset.com
___
--Bandwidth and Colocation Provided by http://www.api-digital.com
Come on you carriers on the list... Give up the dibs what are you using
and why?
About to sledgehammer these SELECT * FROM GARBAGE WHERE SBC = 'nCite'
Don't bother shooting me off Newport Networks stuff... Too pricey
--
J. Oquendo
Joe acquisto wrote:
problem - occasional garbled calls, mostly remote users.
T1 connected to PSTN, SIP over local LAN and internet to remote users. NAT
at local firewall and at remotes. There is no traffic shaping in place, no QoS. Most
are Polycom phones, two Aastra's.
Start with QoS on
QoS does nothing for you unless you're using MPLS between connections
to a degree. (re-stated...) If you're under the impression that you're
going to magically place some auto-qos of sorts and your traffic will be
magically shaped for high performance, you're semi-mistaken. While it
may shape
Any reply ?
2007/7/1, Olivier [EMAIL PROTECTED]:
Thanks everybody for your input.
Let me summarize localization process :
1. Buring boot, phones download from TFTP server an xml or older .cfg file
in which a localization parameter is set.
2. When this parameter is read, phones will then ask
Hi!
I have the same phone with the same problems:
1. Asterisk box does not have fixed IP address, but dyndns name.
2. Phone is at a different location, connected to a router/ADSL modem
Siemens Gigaset (with option not to disconnect from internet ever - set
on).
3. Inside asterisk LAN, phone
Buddies,
Here is my test case
softphoneA--proxyA---Asterisk--proxyB--softphoneB
|
softphone C
softphoneC is registered with Asterisk.
I can place call from softphoneA to softphoneC,and also can make calls
from softphoneC to
We use NexTone for our SBC's on our network. We like:
- 10,000 concurrent calls with media routing
- SIP H.323 signaling with ability to take care of odd vendor
specific issues
- Basic routing engine allows you to create calling plans for
individual end points
- Limits by bandwidth or
We have a setup running Asterisk interconnected to a Panasonic TDA200.
The Asterisk server has a two port E1 card, one connected to the phone
company and the other to the Panasonic. Everything is running fine and
we can send and receive calls from the Panasonic and phone company. We
are
Ade Vickers wrote:
Hi Richard,
Thanks for those replies - I'll give them a shot shortly.
That's not really what I meant by configuration -- you can choose the
MOH source for Asterisk. It's only the native player that restarts the
music file every time someone is put on hold.
We're still
Joe acquisto wrote:
QoS does nothing for you unless you're using MPLS between connections
to a degree. (re-stated...) If you're under the impression that you're
going to magically place some auto-qos of sorts and your traffic will be
magically shaped for high performance, you're
J. Oquendo wrote:
Karl J. Vesterling wrote:
Actually, I *NEED* to change the caller ID. Here's why...
CID internal and external are two different things.
I think Karl was referring to external caller ID.
If PSTN gateway providers lock the callerid to my DID and I have no
way to change
Hi, I don't explain very well what my problem, but i can't make calls.
i analise my log full and i found two errors
Jul 3 19:02:08 ERROR[4670] res_config_mysql.c: MySQL RealTime: Failed
to connect database server on (err 2002). Check debug for more info.
Jul 3 19:02:08 VERBOSE[4670]
Sorry,my mistake,I did not add proper context in the trunk setting of proxyA.
On 7/3/07, Jason Ma [EMAIL PROTECTED] wrote:
Buddies,
Here is my test case
softphoneA--proxyA---Asterisk--proxyB--softphoneB
|
softphone C
Am Donnerstag, den 28.06.2007, 07:07 +0200 schrieb Adam KOSA:
Hi guys,
sorry for the long e-mail, i'm only trying to give as much information
as i think is relevant to my problem (console log, sip.conf and
extension.conf parts). I've sent this e-mail a couple of days ago, but
it bounced
. . .
QOS across the internet is pointless and further more doesnt really
exist, I would suggest setting qualify=200 in sip.conf so that asterisk
will not send a call to the remote end if they are more than 200
milliseconds away.
Away, in what sense? Are you referring to packet
You should make sure it's NI2 for swithctype, and the same goes for
the Panasonic
On 7/3/07, Carlos Chavez [EMAIL PROTECTED] wrote:
We have a setup running Asterisk interconnected to a Panasonic TDA200.
The Asterisk server has a two port E1 card, one connected to the phone
company and
On Tue, 2007-07-03 at 17:18 +0800, Jason Backshall wrote:
Hi Grigoriy,
If I'm interpreting your call flow correctly, you're only ever making one
outbound call - that's the call from ext100. You're then transferring that
call to ext200. Is this correct?
If so, given that you're only making
On Tue, 2007-07-03 at 14:54 -0400, C F wrote:
You should make sure it's NI2 for swithctype, and the same goes for
the Panasonic
We are using R2 (Unicall) signalling not ISDN between the TAD and
Asterisk.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Andy Brezinsky wrote:
We use NexTone for our SBC's on our network. We like:
- 10,000 concurrent calls with media routing
- SIP H.323 signaling with ability to take care of odd vendor
specific issues
- Basic routing engine allows you to create calling plans for
individual end points
-
Anthony Francis wrote:
QOS across the internet is pointless and further more doesnt really
exist, I would suggest setting qualify=200 in sip.conf so that asterisk
will not send a call to the remote end if they are more than 200
milliseconds away.
Oh man if I did that for my homebased ATA
Stephen Bosch wrote:
I ask that you treat people respectfully on the list. The poster has a
valid point and does not deserve that kind of response.
It's possible to disagree and still be civil, and I've no doubt you're
able to do it.
Thanks,
Right sorry list for living in a place called
Joe acquisto wrote:
. . .
QOS across the internet is pointless and further more doesnt really
exist, I would suggest setting qualify=200 in sip.conf so that asterisk
will not send a call to the remote end if they are more than 200
milliseconds away.
Away, in what sense? Are you
On Fri, 2007-06-29 at 12:06 +0200, nik600 wrote:
Hi
how can i retrieve the call unique id of asterisk in the dialplan?
I have enabled the cdr logging on a postgres database.
In the table cdr each record has a field that assumes an unique id
(for example: 1141628669.51)
Can i retrieve
Keep in mind that this law is proposed by the Senator who thinks the
Internet is a series of interconnected tubes which can get clogged.
What did you expect?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users
Are you sure calls were dropped with change in IP ?? I think it should let
current calls run and use new IP for new connections . However if
destination serv drops calls then it's a different story .
On 03/07/07, Henry J. Cobb [EMAIL PROTECTED] wrote:
Asterisk 1.4.5 full log:
[Jul 2 09:31:16]
Think about voicesense which will sense what you are talking and pop in a
*relevant* voice ad to spice up conversation :P .
On 03/07/07, Dean Collins [EMAIL PROTECTED] wrote:
Ooops did Google just become a carrier :)
http://googleblog.blogspot.com/2007/07/all-aboard.html
I hear stocks
. . .
Away, in what sense? Are you referring to packet latency? How does
Asterisk measure this? Ping response?
No, it does NOT measure packet latency. qualify= measures the response
time of the remote device to a SIP OPTIONS packet. If the device is
busy doing something and does not
On 7/3/07, mlists [EMAIL PROTECTED] wrote:
Keep in mind that this law is proposed by the Senator who thinks the
Internet is a series of interconnected tubes which can get clogged.
ommm, isn't that conceptually what a DoS attack is?
___
--Bandwidth
On Tuesday 03 July 2007 7:20 am, J. Oquendo wrote:
(again) Dell. We know based on someone's accent and lack of proper
use of grammar, they are not speaking to us from a location in
the USA. How can we validate that such instance is illegal. It
You obviously have not been around any city centre
I have 2 servers trunked IAX, one of them has 2 TDM800P cards to terminate
calls to PSTN. the problem is all calls to PSTN is almost one-way voice. the
voice is always broken. ztmonitor shows that most of the time of the call
the Tx is zero, while there is always Rx activity on this channel.
Any
Tzafrir Cohen wrote:
[...]
Problems at the zaptel level.
cat /proc/zaptel/*
cat /etc/zaptel.conf
Here the outputs:
[EMAIL PROTECTED]:~# cat /proc/zaptel/*
Span 1: WCTDM/0 Wildcard TDM2400P Board 1
IRQ misses: 10
1 WCTDM/0/0 FXOKS (In use)
2 WCTDM/0/1 FXOKS
Hi List;
Where I determine the codec to be used for the SIP
Trunk (between Asterik and another SIP softswitch)?
Regards
Bilal
Be a better Heartthrob. Get better relationship answers from someone who
Dear List;
To have better security, how can I put a password on
the international calls (if the user dialed the
international call, then it will be asked for password
to send the call outside)?
Can this password read from the CDR file to know whom
did these international calls (using which
Damn!!! Beat me to it ;-}
As an Englishman now living in New Jersey (strangely nowhere near an
exit) I have to say that the local idiom and accent leaves a significant
amount to be desired.
Terms like New Joisey, Shuwa ,wadder, badderies,
congradulations etc make me wonder if I'm in an English
On Tue, 3 Jul 2007, bilal ghayyad wrote:
Where I determine the codec to be used for the SIP Trunk (between
Asterik and another SIP softswitch)?
Are you asking positively how to determine which codec is being
negotiated between those two elements, or, normatively, which one is
best to use?
Hi List;
How can I convert some digits to another digits, and
how I can insert in the end or in the begining some
digits, for example:
If I have a number like 11336784888, then I need to
replace each digit of value 1 by 5, how?
Also how can I add digits to the numbers like adding
00 in the
you might use
sip show peer peername
to see what a peer will allow or
show channel channelname
(channel name as retrieved from show channels)
to determine what a current conversation is using
Moj
bilal ghayyad wrote:
Hi List;
Where I determine the codec to be used for the SIP
Trunk
Bilal,
On Tue, 3 Jul 2007, bilal ghayyad wrote:
To have better security, how can I put a password on
the international calls (if the user dialed the
international call, then it will be asked for password
to send the call outside)?
Asterisk can interface either with a database such as
I'm using Asterisk 1.4.5 (will try 1.4.6 on Thursday, but I don't see
anything in the changelog after the 1.4.5 release dealing with
distinctive ring), zaptel 1.4.3, and wanpipe 2.3.4-10 with a Sangoma
A200 card. I enabled usedistinctiveringdetection in zapata.conf.
However, on the Asterisk
On Wed, Jul 04, 2007 at 12:05:28AM +0200, Administrator TOOTAI wrote:
Tzafrir Cohen wrote:
[...]
Problems at the zaptel level.
cat /proc/zaptel/*
cat /etc/zaptel.conf
Here the outputs:
[EMAIL PROTECTED]:~# cat /proc/zaptel/*
Span 1: WCTDM/0 Wildcard TDM2400P Board 1
thanks for reply. I've same setup with siml. incoming calls 10-12 it works
fine but some time my machies get hang and gives same IAX max data space
error.
thanks
On 6/27/07, Jared Smith [EMAIL PROTECTED] wrote:
On 6/27/07, Arun Kumar [EMAIL PROTECTED] wrote:
so , how much bandwidth I need
bilal ghayyad wrote:
Also how can I add digits to the numbers like adding
00 in the beginning, so the dialed number becoming:
NUMBER=11336784888
exten = 1000,n,Set(NUMBER=00${NUMBER})
0011336784888 or adding digits (like 99) in the end of
the dialed number, so it will become:
I think he means will he be able to tell what passwords were used by
looking at the cdr, for planning purposes, and who used the passwords?
Bilal, using the CDR you could, of course, know who dialed international
numbers. The password, however, will not make it back there unless you
make sure
I've been poking for the definition of QueueMemberStatus and all the
source file indicates is that it is a integer member of the member
structure.
Anyone know where I can find the CONSTANTS definitions?
--
Warm Regards,
Lee
___
--Bandwidth
Contrary to the opinions of Anglo-Philes, we, here in the Colonies,
speak American, not English. In some places, 'Murican.
We get to do that, because, back in the late 1700's . . . we won.
It is only referred to as English out of a sense of compassion.
Oh, so anyway, who was guy Eng you named
Hi,
It looks like your configuration file zapata.conf syntax is wrong. Have
a look in the sample files how to set it up correctly, and if you are
still having troubles, paste your zapata.conf here.
Cheers,
Dimitri
[EMAIL PROTECTED] wrote:
Hi,
I have put Digium TE120P card in PCI slot.
I am thinking of building an Asterisk PBX, and had a question on a piece of
hardware support. I want to include a 4 port PCI 10/100 Switch router card.
For those not familiar it's a PCI card that acts as a switch. My question is
would I be able to configure those 4 ports to support sip phones
On Tuesday 03 July 2007 9:47 pm, Joe acquisto wrote:
We get to do that, because, back in the late 1700's . . . we won.
Hey man, I'm Canadian... We've got our own set of funny accents, and don't get
us started on the Quebecois. Not even the Parisians can understand
THEM! :-)
-A.
J. Oquendo wrote:
Stephen Bosch wrote:
I ask that you treat people respectfully on the list. The poster has a
valid point and does not deserve that kind of response.
It's possible to disagree and still be civil, and I've no doubt you're
able to do it.
Thanks,
Right sorry list for
If it is one of the ones I am familiar with it's only one ethernet
interface and it's literally a switch on a PCI card. The system sees one
interface and there are 4 ports out the back.
If this is the case it's not really instead of a switch so it will
work fine.
-Jonathan
Goran Donev
Change it to ISDN. There is no point in not to, what card do you have
in the TDA200? A PRI or or just T/E1? Since it's too differenct cards
on the TDA200. In fact accroding to Panasonic CallerID isn't supported
on none PRI, although some have gotten it to work.
On 7/3/07, Carlos Chavez [EMAIL
They asked me if we want NI1 or NI2 ?? Is this a reference to the PRI ?
Yes, and just to complicate matters further, they are probably asking about
the NT-1 or NT-2 which is the Network Termination type. NI1/NI2 usually
refers to the National ISDN phase, for which the difference has generally
I have a X100P and a TE110P in my Asterisk box. I can either get the
X100P or the TE110P to work, but never both. Here's my zaptel.conf
span=1,0,0,d4,ami
em=1-24
fxsls=25
When I load wcte11xp and wcfxo, I will get this error.
[EMAIL PROTECTED] etc]# modprobe wcte11xp
ZT_CHANCONFIG failed on
When I upload a pre recorded wav file using trixbox, it can't be played on
the welcome message. But when I record using xlite, it works ok.
trixbox required 8Khz PCM 16bit recording, I used it, but still no success.
Any idea?
Thanks.
Lito
___
Dear all
I have install asterisk 1.2.x and it is working fine my setup is
like
[*]---[Mediant2k][Avaya]
Now i want to transfer call in internal extension i have read more document on
www.voip-info.com but it is now so much clear so if u have any sample
Dear all
I have configure asterisk with avaya so now i have configure
11 for trunk line to goes on avaya system but now i want to replace it with 0
means
I press ' 0 ' it will convert my digit in 11 automaticaly is there any
dialplan to do this ??
Regards
satish patel
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