We are having an issue with the SPA962/932 combo where the phone and the
sidecar will reboot unexpectedly – could be onhook, could be on a call, doesn’t
seem to matter. I read that certain early firmware revisions could cause this
so I’m running what was a week ago the newest available
We are having an issue with the SPA962/932 combo where the phone and
the sidecar will reboot unexpectedly – could be onhook, could be on a
call, doesn’t seem to matter.
I've had no problems at all with my SPA-962/932 combo, and I've used all
kinds of different firmware versions. If I had
button usage?
Thanks,
marty
There is a 5.2.2 firmware available now, but the changelog for it isn't
helpful at all.
PaulH
On Wed, 2007-12-12 at 11:22 -0700, Marty Mastera wrote:
We are having an issue with the SPA962/932 combo where the phone and
the sidecar will reboot unexpectedly
I’ve been tearing my hair out trying to get a Polycom phone (behind a NAT) to
register to an * box behind a Cisco SIP ALG. With known good credentials
configured on the phone and in *, I get 403 Bad Auth when trying to register.
If I put the phone onto the same LAN as * it works fine without
Polycom's were simply not originally built for multi location VoIP.
There
is no NAT support in the Polycom's. We have several networks, being an
ISP,
and have found that when transversing one network say 192.168.2.x with
the *
box on a 192.168.1.x the polycoms were able to communicate
26, 2007, at 12:07 AM, Marty Mastera wrote:
Any recommendations on an economical layer 3 switch? Preferably
something that you have hands on experience with connecting to IP
phones with attached PCs? Specifically I need the ability to set
the VLAN in the phone to tag voice packets
The only reason to route the voice VLAN is if you need the phones to access the
Internet and/or vice-versa. If you only need to worry about the computers on
the data VLAN accessing Trixbox's web interface, I would suggest using the
Ethernet VLAN capabilities of Linux. You can create eth0.vlan1
Any recommendations on an economical layer 3 switch? Preferably something that
you have hands on experience with connecting to IP phones with attached PCs?
Specifically I need the ability to set the VLAN in the phone to tag voice
packets and to set a native VLAN on a per port basis on the
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Miller
Sent: Tuesday, March 06, 2007 7:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Questions
Dave Fullerton wrote on 3/6/07 9:33 AM:
I'm trying to allow
access to an individuals mailbox by having them dial their own DID, wait for
their voicemail greeting and pressing * (to be followed by a password
prompt).
For some reason I
thought that this functionality was built-in to Voicemail but must not be since
it doesn't
Kerry,
We are also a Cbeyond partner focusing mainly on SIPconnect, I thought I
would chime in b/c we don't have that problem setting outbound callerid.
It's true we can't set it to a number not on the customer's account, but
we can set it to any number on the account including DIDs. We do this
If you're trying to check on a particular call, you can do
a 'sip show channel xxx' to display a bunch of info...look for Audio IP which
will tell you where the audio is coming from for a particular
call...
Marty
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philippe
Does anyone have the polycom soundpoint ip's successfully remotely
provisioning? I've got the phone pulling default configs, and it's
downloading phone specific information, but it's not actually using
that
information. Any help would be appreciated :)
--
Aaron Daniel
Aaron,
I had
Slightly off topic as this doesn't pertain directly to
Asterisk, but with the Polycom 500/501 phones, does anyone
know how to correctly put a custom logo for the idle screen
on the device? I've read the Admin Guide through and through
and the information there is not enough to implement
of a call (a hangup), but we are
troubleshooting bad audio in both locations and the wording of these messages
doesn't appear benign.
thanks
Marty Mastera
M3 Resources
[EMAIL PROTECTED]
Phone: 303.680.1283 x200
FAX:
206.666.1786
___
Asterisk-Users mailing
Grasping at straws
here...is anyone using a Dell PowerConnect 2224 24-port unmanaged 10/100 switch
in a deployment?
I have two separate
asterisk installations with bad one-way audio where the only common elements
left are the Dell switches and Polycom IP-500 phones. Two different ITSPs
-
Steven Langley wrote:
Hi
there
I have a question regarding IAX
jitter. I have 3 users on a LAN dialing into a Meetme conference on an
Asterisk box which is also hosted on the LAN. I have set jitterbuffer = no
and tos = lowdelay.
Hi Marty -
The complaint from the users is that calls cut out,
kinda like when
you have spotty cell coverage. Doesn't seem to matter whether the
call is incoming or outgoing, although it might be true
that my users
hear the remote party cut out, while the remote party
doesn't
I thought I would
throw this out there and see if anyone has any ideas...I have the same problem
at 2 locations.
The complaint from
the users is that calls "cut out", "kinda like when you have spotty cell
coverage". Doesn't seem to matter whether the call is incoming or outgoing,
although
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Wiley Siler
Sent: Thursday, April 28, 2005 12:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom IP500 - Phone TIme
H my phones are
I'm having a problem
with SIP dtmf negotiation during call setup.Myproviderwants me
to use rfc2833, which I configured in the general section of sip.conf but it's
not working. From packet capture and sip debug we see that my provider is
offering 0 and 105 (0=ulaw, 105=a codec used on their
I have setup my system to give a company announcement if
somebody calls, ...
I would like to avoid these announcements, if the caller is
known by the system.
Each caller I would like to put into a database with name.
Now we know them!
If we know them, we do not announcement.
Is
Jason Brown wrote:
| Anyone have experiece with polycom phones?
|
| I am experiencing a really weird problem. In an office
where I have
| the following extensions:
| On the Polycom phones, when I want to dial from extension
100 to any
| extension 120 or above, or dial out, it
Jason Brown wrote:
| Anyone have experiece with polycom phones?
|
| I am experiencing a really weird problem. In an office where I have
| the following extensions:
| On the Polycom phones, when I want to dial from extension
100 to any
| extension 120 or above, or dial out, it dials
In our setup, outbound call volume frequently exceeds the
line capacity of the DSL line. We do not want to move to
another codec to better utilize the line, but instead wish to
automatically divert overflow to the Long Distance T1 when
the DSL is full. Ideally the system would also be
Hi
all,
Got an interesting question
here.
We have 2 incoming ISDN channels
(linked to a local phone number that our customers call), and an account with
a local IP provider. We use the ISDN for local, 13 calls 13 is a call whos
destination depends on the
map that I want to display:
bitmap.IP_500.66.name ="arf" but from there I'm not sure where to go...what do
you change to tell the phone to actually use that bitmap on the main screen
during idle conditions?
Thanks
Marty Mastera
M3 Resources
[EMAIL PROTECTED]
Phone: 303.680.1283 x20
Has anyone got this to work? Under Idle Display Animation, the
administrators guide says For example, a company logo could be
displayed..
In the ipmid.cfg file, I enabled 'ind.idleDisplay.enabled'
(ie changed
it to 1), and under the IP 500 section, I added an entry for the
On Asterisk 1.0 with a 4-port Digium FXO card, voicemails
from the PSTN have volume so low they often can't be heard.
Worse, callers sometimes get cut off in the middle of leaving
a message. It is extremely frustrating to hear ...and my
number is...END OF MESSAGE
A search of the
The full text of the bug you reference above indicates that
pstnVMgain
was (or is) part of an ongoing feature request/bug report
and has not
been implemented for use at this time (and may never be).
Right. So -- what can I do to boost volume of PSTN - * voicemail?
thanks
dn
Hello:
I have read that
music on hold requires a timing source (which I never had to worry about
previously since the server had zaptel hardware in it)...now I'm configuring a
server in a colo which has no zaptel hardware.
If I use the
dialplan to run MusicOnHold(), I do get the music
Hmmm... I have this aweful feeling that I'm choosing the
exact wrong time to ask a newbie question :) Oh well, here
it goes.
The quick question is : How do I dial an extension?
(answer is probably - you don't in which case:) How do I
dial my asterisk box? - I have no outside
incoming and outgoing calls. Does this seem
reasonable? If any of you are accomplishing this in other ways, please
share your examples...
Marty Mastera
M3 Resources
[EMAIL PROTECTED]
Phone: 303.680.1283 x200
FAX:
206.666.1786
___
Asterisk-Users mailing
on how the IP-10 is working these days with SIPif it's reasonably
solid it's probably at the top of my list...
Thanks
Marty Mastera
M3 Resources
[EMAIL PROTECTED]
Phone: 303.680.1283 x200
FAX:
206.666.1786
___
Asterisk-Users mailing list
Asterisk
for approx. 10 users. Next logical
question: what other phones would you recommend for a situation like this (built
in switch, display, speaker phone...)
Thanks
Marty Mastera
M3 Resources
[EMAIL PROTECTED]
Phone: 303.680.1283 x200
FAX:
206.666.1786
to the right location for
it?
Thanks,
Marty Mastera
M3 Resources
[EMAIL PROTECTED]
Phone: 303.680.1283 x200
FAX:
206.666.1786
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
Matthew:
That feature is referred to as Answer Supervision, and the dial flag is c...beware
however, it only works when the outbound call is made via a Zap channel.
I have the exact same need as you, however the call is made using either sip or iax,
and currently answer supervision isn't an
I haven't seen any
status on the 7960 backlight project lately...I tried to email the original
poster but his mailbox appears to be over quota. Does anyone have an
update on this?
Thanks,
Marty Mastera
M3 Resources
[EMAIL PROTECTED]
Phone: 303.680.1283 x200
FAX:
303.680.1283
IAXTel
When I logged into Tech Data this morning, the PAP2-NA was
now marked as discontinued and no longer available and only
the PAP2 version was available which is the Vonage branded version. :(
I saw someone on the list say that they heard from Cisco that
these units were not due out until
em happens. Dialing very slow and deliberate seems to
help, although I haven't done super serious testing of that
yet...
Any
ideas?
Marty Mastera
M3 Resources
[EMAIL PROTECTED]
Phone: 303.680.1283 x200
FAX:
303.680.1283
IAXTel: 700.206.7507
F
Please explain how you got the PAP2 to work with another
carrier? I spent over an hour on the phone with 3 levels of
Linksys support staff and 2 levels of Vonage staff telling me
that the PAP2 CAN NOT be used on any other service except
vonage because they burn the vonage information
I had 2 senior level management people at linksys corp
confirm that this would not be possible until December. They
both told me that they are currently in development of a
'non-locked' version but that it would not be in stores until
December.
Did you find these PAP2-NA at Fry's as
een
much (success or failure) on the list from anyone attempting fax over
IAX.
Thanks
Marty Mastera
M3 Resources
[EMAIL PROTECTED]
Phone: 303.680.1283 x200
FAX:
303.680.1283
IAXTel: 700.206.7507
FWD: 484162
___
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[EMAIL PROTEC
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William
C. Lohr Jr.Sent: Tuesday, September 14, 2004 10:07 PMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Sending
Caller ID info in MD/USA
All,
Having trouble getting answer
from Verizon. I
All,
Having trouble getting answer
from Verizon. I believe Asterisk will let me specify a name and number
that is sent to the PSTN (Verizon) of outgoing calls. For instance, if I
have a client, First Bank, and their toll free number is 888-555-1234, I could
send that
I've tried setting up my sip.conf in two ways:
--
register = [240xxx]:[EMAIL PROTECTED]
[Broadvoice]
type=peer
username=[240xxx]
fromuser=[240xxx]
secret=[my_password]
host=sip.broadvoice.com
context=incoming
Im trying to get IAX to work between my * and FWD. I
activated my iax2 account on iax.fwdnet.net and I get the output:
Registered to '65.39.205.121', who sees us as 68.14.203.254:4569
when I start asterisk. I tried used the Call Me tool on
fwdnet.net but I dont get any calls even
Are the FXOs on the 2x on ports 1-2 or 3-4? Maybe it has to
do with *any* FXO on port 1...
Please get back with the list with your findings.
My experience led to a replacement from Digium, but the card is a
TDM400P with 4 FXO...now that I think of it, during troubleshooting
there
Works for me, follow the instructions closer. :)
Storm D. J. Petersen wrote:
Hi,
I cannot seem to accept incoming calls from FWD using IAX2.
I followed the
directions posted at www.fwd.pulver.com/advanced/iax I can
make outgoing
calls fine using IAX via FWD. When someone calls
n anyone recommend either an alternative
solution or a starting point for implementing it in the code of other
channels?
Thanks,
Marty Mastera
M3 Resources
[EMAIL PROTECTED]
Phone: 303.680.1283 x200
FAX:
303.680.1283
IAXTel: 700.206.7507
FWD: 484162
__
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tim Robinson
Sent: Wednesday, September 08, 2004 1:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Answer confirmation on non-Zap channels?
Hi
I wonder if anyone else's Asterisk
box drops the connection to voicemail after 30 secs even when the maxmessage
parameter is set to 180 (3 mins). Here is the general section of my
voicemail:
Roger,
There has
been very recent discussion regarding this topic
I'm using RC2 and last weekend's changes from VoicePulse. Outbound
calling and dtmf works fine. However, an inbound call to my DID
cannot
send dtmf digits to the IVR.
Thoughts?
I have the same problem...my iax.conf is set up exactly as recommended
per the recent Voicepulse changes and
Since Thursday evening my asterisk box has been failing to register
with
broadvoice. I haven't changed any of my config files in the last week.
Can anyone suggest anything?
Asterisk is reporting:
*CLI Aug 28 16:15:17 NOTICE[6150]: chan_sip.c:3914 sip_reg_timeout:
Registration for
I had the same problem. To fix it, I had to do two things
First: I had to update to CVS head, this was as per broadvoice
support.
Second: After updating, I had to change my sip.conf. Originally my
sip.conf used hard coded ip addresses for broadvoice's IP servers, so
I
had to
I had the same problem. To fix it, I had to do two things
First: I had to update to CVS head, this was as per broadvoice
support.
Second: After updating, I had to change my sip.conf. Originally my
sip.conf used hard coded ip addresses for broadvoice's IP servers, so
I
had to
Well, I had the TDM act strange this afternoon and spontaneously drop
the
line. And as I expected, it locked up this evening. It seems to have
hung
the machine.
This system has not gone more than 4 days without a soft boot since I
put
the
TDM400 into it. I've had it for 3.5 years and
Is anyone aware of a single FXS ATA with a built-in switch ie
2 LAN ports (other than the Cisco ATA 188)?
Grandstream 486 users: is it possible to disable the router/NAT
functionality and configure the WAN port as switched with the LAN port, effectively
giving you two switched Ethernet
Sorry for the very newbie-like question. I have the
FXS part straight. The part I don't understand is the
FX0 part. Will I need the FX0 card if I am connecting
to a service like FWD? My goal is to get rid of my
phone line all together. I am under the impression I
will only need an FX0
Hello everyone,
I playing around with ENUM and have configured * to query a
few sources for testing purposes (fierymoon, e164.arpa, e164.org). Id
like to know if there is a way to query these servers manually (ie outside of
asterisk via nslookup or equivalent) to find out if particular
I have also been having problems today registering... I contacted
them,
but
they have no known issues. It finally did register on it's own.
For those having trouble with BV, try this for a test: in sip.conf,
replace occurrences of sip.broadvoice.com with this IP: 147.135.0.129,
Eats humble pie!!
I'd never seen it in the settings and sure enough it's there.
Sorry for misguiding.
P
No need for that!...thanks for the info everyone...I'm going to start
keeping my eyes open for a WRT54G for a good deal somewhere!
Thanks again,
Marty
What I am NOT able to do is dial a seven digit local or 10
digit long distance number and make a phone call to the pstn
using the x100p card.
snip
Attached is my extensions.conf
When I dial 94341321 or 4341321 I just get a 404 error in Xlite.
What am I doing wrong? Any help would
Thanks for the tip, that made things work, it is really
difficult for me to understand the different config files and
especially the extensions.conf, it is very confusing. I am
trying to learn though.
Now that I have got outgoing calls to work from the sip
phone. How can I route
When I call the pstn number, the zaptel picks up the line on
the first ring and then forwards it to the sip phone and
rings it. Is there anyway to prevent the zaptel from picking
up the line until the sip phone actully answers the call.
This way I could answer the phone either locally on a
Hello everyone
Searching the archives and google always comes up with entries regarding
the dyn dns option in the 7960, but I can't find answers to my
specific question
My 7960 is connected via cable modem and is NAT'ed (everything is
working fine). On the 7960 under SIP
It's a way to specify a DNS via config file which has
priority over whatever is handed out from DHCP.
(Optional) IP address of a new dynamic DNS server. If a new
DNS server address is specified, it is used for any further
DNS requests after the phone uses the initial DNS address
upon
If I understand you correctly, you are trying to figure out why you must
dial ext 142 prior to dialing exten 666 in order to get the 7960 to
connect you with 666 - please forgive me if I misunderstood...
Anyway, it appears as though the two contexts you have listed below have
the exact same name
Greg,
Per your suggestion, I added dtmfmode=inband to the general section of
my sip.confthe other items you mentioned were already in sync with
what I had. With that one change inbound DTMF to * IVR works!
I will continue to play with it to flesh out it's reliability, but I was
successfully
Hello everyone,
Having recently gotten Broadvoice inbound DTMF to work (thanks Greg)...I
am now running into a frustrating problem...when a call comes in to the
BV number via a cell phone (tested with 3 different cell phones; albeit
all on T-Mobile) the beginning of the IVR welcome audio is cut
Jay:
I hope this input is in some way helpful, if only to confirm your
findings...
I don't have a Sipura to play with, but I do have a X100P...I used my
cell phone to call my BV number, then answered the call from my 7960 and
transferred the call out the X100P to my cell providers voicemail
I can verify this with a similar arrangement...
Normal outbound calls = good DTMF
Normal inbound calls = no DTMF processed
If I place a call to my cell phone via BV, then transfer the call to an
internal extension which puts my cell phone into an IVR menu, DTMF
presses from my cell phone are
Hello
everyone,
I have contacted
providers listed on the Wiki and other lists of providers offering SIP or IAX
service...
So far I'm having
trouble finding an IAX provider offering the 303 or 720 area codes. Can
anyone suggest a IAX provider that offers these areas codes? I would
prefer
(as far as I
know), so I realize that I would not be able to configure it in this case
and would have to have them do it.
Do this sound accurate?
Thank you,
Marty
On 10 Jul 2003 20:58:13 -0500 Steven Critchfield wrote:
On Thu, 2003-07-10 at 18:33, Marty Mastera wrote:
Hello,
I don't
phone be
recommended (with transfer, hold, callerid, etc...capabilties)?
Thanks again Andy,
Marty
On Thu, 10 Jul 2003 23:02:22 -0500 Andy Hester wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Marty Mastera
Sent: Thursday, July 10, 2003
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