Hi,
I have a Digium TE410p T1 card and I've noticed that under asterisk
1.4.17/18 I have problems detecting DTMF in IVRs. I think I've
narrowed the problem down to some sort of interference between the
greeting that is playing and the DTMF tones. DTMF detection seems to
work very reliably when I
Hi everyone.
I've tried RelaxDTMF and it didn't seem to help.
I can't use Read() or WaitExten() instead of background.
What I was doing was running a test using the following dialplan:
[custom-testdtmf2]
exten = s,1,Answer
exten = s,n,AGI(festival-script.pl|Enter test digits. Then press pound.)
Another note, after looking at the source code, it seems as though
WaitExten() and Read() use ast_waitfordigit() where Background uses
ast_waitstream().
Apparently these 2 functions must behave differently.
-- James
On Wed, Mar 12, 2008 at 9:45 AM, James Lamanna [EMAIL PROTECTED] wrote:
Hi
behave differently.
-- James
On Wed, Mar 12, 2008 at 9:45 AM, James Lamanna
[EMAIL PROTECTED]
wrote:
Hi everyone.
I've tried RelaxDTMF and it didn't seem to help.
I can't use Read() or WaitExten() instead of
background.
What I was doing was running a test using
Hi,
Is it possible to get the XML config off of a Linksys SPA-941 or 942 phone?
I've tried http://[ip address]/admin/spacfg.xml however that file
doesn't appear to exist.
Thanks.
___
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Hi,
I've noticed that sometimes instead of getting a regular ring tone
when calling out on a Zap channel, I get this obnoxious loud noise
which forces me to hang up.
Is this a problem in the Zaptel driver? I seem to recall that ringback
tones are generated by zaptel when dialing out from a SIP
you get over a zap channel (be it
analogue or digital) is generated by the remote end, /not/ Zaptel.
The ringback you get over a SIP or IAX2 channel is often generated by
either Asterisk or the SIP/IAX2 device you're calling from.
James Lamanna wrote:
Hi,
I've noticed that sometimes
Hi,
Is there any information that can be gathered from the logs about why
a SIP call was dropped/terminated without either side hanging up?
I've run asterisk pretty verbose and I guess I haven't seen anything
that pops out at me yet.
I'm trying to diagnose why some clients are getting dropped
Hi,
While I haven't personally used any of their equipment yet, Brix is
supposed to have good h/w and software for measuring a MOS score:
http://www.brixnet.com/products/BrixCall.shtml
http://www.voiptroubleshooter.com/basics/mosr.html
-- James
Hello Fellow Users,
I am looking for a way -
Hi,
A couple of our users are reporting that intermittently, their
voicemails are unable to be heard because there is a
milliwatt-sounding tone recorded over the top of it.
Has anyone else encountered this issue?
I have put a recording of the voicemail up online for people to listen
to to see what
Using a frequency analyzer, the tone is composed of 1Khz multiples at
(1, 2, 3, and 4Khz).
Any ideas?
On Mon, Jun 30, 2008 at 2:46 PM, James Lamanna [EMAIL PROTECTED] wrote:
Hi,
A couple of our users are reporting that intermittently, their
voicemails are unable to be heard because
Here are more specifics that I forgot to include:
- Asterisk version is 1.4.18.
- Call was coming in over a PRI provided by PacWest through a Digium TE410P.
- ulaw codec.
Thanks.
-- James
-- Forwarded message --
From: James Lamanna [EMAIL PROTECTED]
Date: Tue, Jul 1, 2008 at 5
Hi,
We have a few Aastra 480ci phones and we've noticed that in order to
get the phone to receive a call, qualify must be = no.
Apparently the Aastras do not respond to the qualify message (or
respond in a way Asterisk doesn't understand) and Asterisk thinks the
phone is unreachable.
However, this
Hi,
Can someone tell me what causes asterisk to Auto-congest a phone on
a SIP channel?
Is it just a lag issue to the phone or is there something else going on?
Thanks.
-- James
___
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Hi,
I'm having an issue where some phones behind a sonicwall are auto-congesting.
The status on sip show peer shows ping times anywhere from 80ms all
the way up to 1100ms.
PCs behind the same firewall have a ping time of about 30ms to the PBX itself.
Does anyone know if the sonicwall is inserting
, James Lamanna [EMAIL PROTECTED] wrote:
Hi,
I'm having an issue where some phones behind a sonicwall are auto-congesting.
The status on sip show peer shows ping times anywhere from 80ms all
the way up to 1100ms.
PCs behind the same firewall have a ping time of about 30ms to the PBX
itself
Hi,
I've noticed that if I have a multi-line linksys (942 or 962) phone
with the same sip registration mapped to each line key, that if all
the lines are full the phone will accept another call. I would expect
the phone to respond with busy so the call would to directly to
voicemail.
Has
with different numbers
of lines, I really do not want to do this manually for every extension
I have.
-- James
2008/12/5 James Lamanna [EMAIL PROTECTED]
Hi,
I've noticed that if I have a multi-line linksys (942 or 962) phone
with the same sip registration mapped to each line key, that if all
Hi,
A couple of our customers are having issues with doing attended transfers.
What happens is Caller A receives a call, they transfer to Caller B,
tell Caller B who is calling, etc.. and then
hit the Transfer key again to transfer the call.
Caller A's side hangs up as expected, but the call is
Hi,
I've been trying to use the FaxGateway application to send T.38 out
over Zaptel using asterisk but I don't seem to be having any luck.
I'm executing it in the dialplan like: FaxGateway(Zap/g0/[number])
Has anyone had any luck using this thing and can enlighten me on how
it's supposed to be
Hi,
Here's part of the log that I see.
In this case I'm testing on a box that unfortunately doesn't have a
PRI connection.
I've so far tested with just voice calls so far, but as you can see,
FaxGateway can't even dial out to the SIP trunk properly.
Here's also what the dialplan looks like:
Hi,
I'm trying to get T.38 passthrough to work to T38Modem and Hylafax so
I can terminate T.38 faxes from an ATA.
However I haven't had much luck. I've tried two supposedly T.38
capable ATAs, the Grandstream 206 and the 502.
The 286 seems to connect to T38Modem, but Hylafax doesn't get any fax
Hi,
Using rfc2833, I constantly have the problem that if a user presses
digits reasonably fast,
I will see a bunch of DTMF end emulation of 'X' queued on SIP/.
This queuing screws up the DTMF because the digits never get sent!
Is there a way to prevent asterisk from queuing DTMF? Or at
Hi,
One of our customers has an issue with the callee not being able to hear them.
It seems to happen very frequently on one number in particular where
there are about 3 IVR menus to dial through
before getting to a live person. However, this does not happen on every call.
Running tcpdump on the
] On Behalf Of James Lamanna
Sent: Tuesday, January 27, 2009 12:38 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Muted sound on a Linksys 962
Hi,
One of our customers has an issue with the callee not being able to hear
them.
It seems to happen very frequently on one number
Hi,
I had a Trixbox 1.4.18 that I yum updated to 1.4.22.1.
Now, I seem to have a huge problem with phones not staying registered
(registrations worked perfectly at 1.4.18).
I phone will register the first time I plug it in, and then once you
make a call and hangup (or sometimes even during the
/104
defaultip=192.168.23.114
mailbox=104
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Lamanna
Sent: Wednesday, January 28, 2009 1:01 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk
-
From: James Lamanna [mailto:jlama...@gmail.com]
Sent: Wednesday, January 28, 2009 1:59 PM
To: asterisk-users@lists.digium.com
Cc: da...@debsinc.com
Subject: Re: asterisk-users Digest, Vol 54, Issue 94
Date: Wed, 28 Jan 2009 13:11:19 -0600
From: Danny Nicholas da...@debsinc.com
Subject
[moving to asterisk-users by request]
On Tue, Jan 27, 2009 at 12:56 AM, John Todd jt...@digium.com wrote:
On Jan 26, 2009, at 7:38 PM, James Lamanna wrote:
On Jan 26, 2009, at 8:53 PM, James Lamanna wrote:
Hi,
Is it just me, or does DTMF queuing not work properly?
I'm consistently faced
Hi,
I'm trying to run Asterisk 1.4.23.1 on a small ARM linux board (TS-7800).
Everything compiles fine, but on startup Asterisk always crashes while
loading chan_sip.
If chan_sip is removed, it starts up fine, but I really need SIP to work.
Any ideas?
Thanks.
-- James
Hi,
I have a couple of users who are having a peculiar problem.
On some outbound numbers where there is a deep IVR tree (3+
selections), and then a live person picks up,
the live person will be unable to hear them on the phone, but they can
hear the live person.
I've done packet traces and it
Hi,
I'm trying to use iaxmodem against a very old version of asterisk
(1.0.7 - its a debian sarge embedded system),
yet when asterisk gets a call from iaxmodem, it says that the format
for the call is unknown.
Does anyone know if there is a minimum version of asterisk that is
compatible with
On Sat, Feb 7, 2009 at 1:19 PM, Lee Howard fax...@howardsilvan.com wrote:
James Lamanna wrote:
Hi,
I'm trying to use iaxmodem against a very old version of asterisk
(1.0.7 - its a debian sarge embedded system),
yet when asterisk gets a call from iaxmodem, it says that the format
On Sat, Feb 7, 2009 at 1:44 PM, James Lamanna jlama...@gmail.com wrote:
On Sat, Feb 7, 2009 at 1:19 PM, Lee Howard fax...@howardsilvan.com wrote:
James Lamanna wrote:
Hi,
I'm trying to use iaxmodem against a very old version of asterisk
(1.0.7 - its a debian sarge embedded system),
yet when
Hi,
Is there a way to do a blind transfer within an asterisk dialplan (like '##')?
The reason I need this (I think) rather than a regular Goto() is that
I'm trying to do one-touch parking.
I can park a call using one-touch parking and then pick it up again,
however if I try to re-park the call, it
Hi,
Does anyone have some good examples of a Kamalio or OpenSips
configuration that integrates with Asterisk?
Essentially I want to use the SIP router as the UA, but still run all
the calls through Asterisk (for dialplan, etc..)
I've looked for examples on the project web sites, but I haven't
Hi,
We've implemented a 'page-all' function for some of our customers, and
we've noticed that
on occasion the page-all will cause asterisk to crash (safe_asterisk
then restarts it again).
The particular customer has about 20 phones, and also has 5 Linksys
932 to monitor the state of these
Hi,
I have a strange problem. At a site where there are 20+ phones, there
is one phone that cannot make outbound (to PSTN) calls.
Each call is dropped after 20s with no response to our critical packet.
Calls to voicemail and internal extensions work fine.
I understand that everything points to a
and please CC me, I'm reading in digest mode..)
-- James
On Fri, May 22, 2009 at 10:36 AM, James Lamanna jlama...@gmail.com wrote:
Hi,
I have a strange problem. At a site where there are 20+ phones, there
is one phone that cannot make outbound (to PSTN) calls.
Each call is dropped after 20s
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 peer on those
extensions shows them as OK.
Therefore, I have no way to tell this
Oliver wrote:
How many phones are concerned ?
The box currently has about 380 active phone registrations.
Thanks.
Please CC me directly as well because I'm on digest mode.
-- James
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Hi,
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.
Thanks.
-- James
On Fri, Jun 26, 2009 at 11:10 AM, James Lamannajlama...@gmail.com wrote:
Hi,
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
On Fri, Jun 26, 2009 at 1:31 PM, James Lamannajlama...@gmail.com wrote:
The use case is that a customer has a fax machine attached to an ATA.
The ATA sends T38 to Asterisk over SIP, then I need to forward that out
the PSTN.
Got it. I'm saying why not skip the ATA and asterisk, and plug the
On Fri, Jun 26, 2009 at 11:10 AM, James Lamannajlama...@gmail.com wrote:
Hi,
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
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?
I have tried that. Unfortunately restart when convenient doesn't
always seem to actually restart
asterisk, presumably because there are stuck calls
Hi,
Some users have been reporting a peculiar problem.
The are having an issue when they dial out to some multi-level IVRs
where you make 2 or 3 touchtone choices and then are connected to a
live operator.
When the live operator connects the operator cannot hear them or
sometimes it results in
Does Asterisk 1.6 fully support RFC4235?
Or is it the same implementation as 1.4?
Thanks.
-- James
___
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now:
Hi,
I'm coming up with ideas about building a cluster of asterisk servers,
and am exploring the virtualization option.
I'm curious to know some real-world data about how many extensions a
VMWare install on good hardware could support.
I've seen stories about how the hypervisor timeslicing can
On Fri, Aug 7, 2009 at 11:47 AM, James Lamannajlama...@gmail.com wrote:
Hi,
I'm coming up with ideas about building a cluster of asterisk servers,
and am exploring the virtualization option.
I'm curious to know some real-world data about how many extensions a
VMWare install on good hardware
.
Please email me directly for more details or any questions about the position.
Thanks.
James Lamanna
Warp2Biz, Inc.
___
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now
Hi,
I'm trying to configure 2 parallel sip trunks between 2 boxes.
However I seem to have the problem that when making a call from Box 2
to Box 1, it sometimes
says authentication failed because it is using the username of the other trunk.
Here's my configuration:
Box 1:
[dp-dp2]
type=peer
Hi,
I have two asterisk boxes AB connected together via IAX.
Phones register to Asterisk box A, and Asterisk box B is the PSTN connection.
When dialing a number from a phone registered to A that DAHDI returns as BUSY,
the Busy(20) application returns immediately instead of playing the busy tone.
Hi,
I noticed that Dahdi starting producing these error messages:
ERROR[29250] chan_dahdi.c: No more room in scheduler
ERROR[29250] chan_dahdi.c: Asked to delete sched id -1???
during which time I could not send any calls or receive calls on at
least one of my Dahdi spans.
The only way to clear
Hi,
I've noticed that my MeetMe install seems to think chan_dahdi is missing:
app_meetme.c: No DAHDI channel available for conference, user
introduction disabled (is chan_dahdi loaded?)
However, it definitely is since I have 3 PRIs functioning normally :)
Is there anything I should check before
Hi,
Has terminating T.38 to PSTN found its way into the asterisk 1.6 mainline yet?
I remember seeing an app_gateway floating around at some point a while
ago, but I never had any luck with it.
Thanks.
-- James
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Hi, I have a PRI problem where it appears that my system is not
responding to SETUP messages on a channel.
It seems to be retransmitting a significant number of RELEASE messages
to clear a call that is most likely
to be long gone.
This causes a huge issue because I get a bunch of hangup cause 102s
On Fri, Feb 12, 2010 at 12:54 PM, James Lamanna jlama...@gmail.com wrote:
Hi, I have a PRI problem where it appears that my system is not
responding to SETUP messages on a channel.
It seems to be retransmitting a significant number of RELEASE messages
to clear a call that is most likely
Hi,
I have a case where SIP channels will not be destroyed, resulting in
further calls to ChanIsAvail() to fail.
The process (I believe) to replicate this is the following:
- Make a call to another SIP phone that is an intercom call (Auto-Answer)
- For whatever reason, the phone happens to go
Hi,
Does anyone have any good empirical data suggesting what the maximum
number of PRI calls (incoming and outgoing)
without hardware echo cancellation can be handled on a single box is?
I have a TE410P T1 (1st gen) card and I'm seeing interesting errors of
D-Channels going down and then coming
Zeeshan A Zakaria wrote:
On Wed, Mar 24, 2010 at 5:42 PM, James Lamanna jlama...@gmail.com wrote:
[snip]
The specs of the machine are, Dual Xeon 2.80Ghz (both single core but w/HT)
4GB memory.
Running asterisk 1.4.26.3 (32-bit)
with libpri-1.4.7 and zaptel-1.4.12.9
So I think it is not your
Hi,
I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall.
After some period of time, asterisk says that some of them are
unreachable, and the phones lose their registration.
The only way to make the phones recover is to clear the NAT
translation tables for the phones on the PIX (clear
that connection so your NAT/firewall won't just
close it.
Sorry, should have mentioned that all these phones have qualify=yes
and nat=yes in sip.conf.
Thanks.
-- James
On Sat, Mar 27, 2010 at 8:17 AM, James Lamanna jlama...@gmail.com wrote:
Hi,
I have about 10 Cisco 7960s behind a PIX 506E (IOS
On Sun, Mar 28, 2010 at 8:28 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Sun, 28 Mar 2010, Joseph Begumisa wrote:
Can anyone recommend a 24 fxs port voip gateway that has worked well with
asterisk? I have a couple of analog handsets that I want to hookup to my
asterisk server? Any
Hi,
I'm trying to figure out the cause of a soft lockup I experienced:
Mar 29 09:38:24 pstn1 kernel: BUG: soft lockup - CPU#0 stuck for 10s!
[asterisk:32029]
Mar 29 09:38:24 pstn1 kernel: Pid: 32029, comm: asterisk
Mar 29 09:38:24 pstn1 kernel: EIP: 0060:[c046e7fe] CPU: 0
Mar 29
SC1425 - Dual, dual-core Xeon Processors.
I'm hopefully going to be able to stress test this machine to see if I
can make it panic again with the PRI card IRQ isolated to CPU0. If so,
I'll see if it does the same thing on the other cores...
-- James
--
Matt
On Mon, Mar 29, 2010 at 8:30 PM, James
On Mon, Mar 29, 2010 at 9:23 PM, James Lamanna jlama...@gmail.com wrote:
On Mon, Mar 29, 2010 at 8:38 PM, Matt Watson m...@mattgwatson.ca wrote:
Dell server by any chance?
I have a similar problem with a TE220B in a Dell 1950 III server - i've seen
several other people having issues
Hi,
I'm seeing a lot of Exceptionally long voice queue length errors in
my logs, and then I seem to have a problem
where Asterisk will drop the registration for a significant number of
phones (they go UNREACHABLE), but then they
come back approximately a minute later.
Is this some sort of load
On Thu, Apr 1, 2010 at 6:15 AM, Jaap Winius jwin...@umrk.to wrote:
[snip]
Besides the above error, I also noticed this:
CLI pri show span 1
Primary D-channel: 16
Status: Provisioned, Down, Active
Switchtype: EuroISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
Hi,
I'm running Asterisk 1.4.26.3 and I've noticed an interesting problem
when trying to play a Busy tone over a IAX trunk from the PSTN.
It seems as though Busy(20) returns non-zero immediately (it does not
wait 20s), so the caller never hears the busy tone, but
the call just appears to hang up.
On Fri, Apr 9, 2010 at 5:17 PM, Danny Dias ing.diasda...@gmail.com wrote:
Hello my friends,
I want to make fax work in the following scenario:
My versions are:
Asterisk 1.4.21.2
WANPIPE Release: 3.4.7
Zaptel Version: 1.4.11
libpri version: 1.4.5
Digium Card TDM 410P
The E1 pri is
On Sat, Apr 10, 2010 at 6:35 AM, Adrian Marsh
adrian.ma...@ubiquisys.com wrote:
Hi All,
I’ve two asterisk servers on the same LAN, both 1.4, and I keep getting “Got
SIP response 489 Bad event back from 192.168.3.10”
No idea whats causing it. The only references I can find mentions NATing
On Sat, Apr 10, 2010 at 8:50 AM, Tarek Sawah tareksa...@hotmail.com wrote:
Greetings list
i'm trying to connect with a VoIP provider for termination.. and they have
offered us three servers to connect with
one SIP Signaling server and Two Media servers ..
googled for a week and didn't find
Hi Jerry,
On Thu, Apr 8, 2010 at 6:54 PM, Jerry Geis ge...@pagestation.com wrote:
I am looking for something in asterisk that
will let me record a wav file in asterisk (which I know how to do)
then some other command (external or dialplan) that would read
the wave file and tell me if a
Hi,
Has anyone had any experience using DRBD to mirror an entire asterisk machine?
If so, is there a performance issue at all when people are recording
voicemails and the like?
It seems like that could generate quite a bit of traffic. Also, do you
bother to mirror the log files as well?
Thanks.
Hi,
After playing around with queues a bunch on 1.4.26.2, I noticed a few things,
which the patch below addresses. It addresses:
- Callers in position 0 will hear periodic/position announcements at a
very different rate than all other callers.
-- Announcements while in position 0 could be
Hi,
I have a duplicated DTMF issue with, it appears, bridged IAX channels.
I have the following setup:
PRI IAX
* PSTN ---* Dialplan
I've configured a number on the dialplan server to make and outbound
call to the pstn. This call then comes back into the dialplan
On Wed, Apr 28, 2010 at 6:57 PM, James Lamanna jlama...@gmail.com wrote:
Hi,
I have a duplicated DTMF issue with, it appears, bridged IAX channels.
I have the following setup:
PRI IAX
* PSTN ---* Dialplan
I've configured a number on the dialplan server
It seems that the PAP2T does support TFTP and an XML-based config for
deployments...
I've used both the Grandstream 286 and the Linksys PAP2T.
I have been able to get some limited faxing to work using T30 with a PAP2T.
Configuration and provisioning of the Linksys is very easy through
either
Hi,
I'm having a problem trying to get a Cisco 7965 phone registered on
Asterisk 1.4.26.
As we know, Cisco now, for security reasons, has made the phone ports
non-symmetric, in that it sends out UDP requests on a high port and
receives them on a different port.
It seems that, even with 'nat' set
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Lamanna
Sent: Wednesday, May 05, 2010 12:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Registering a Cisco
On Thu, May 6, 2010 at 8:14 PM, Vineet Bhojnagarwala vbho...@gmail.com wrote:
Alternatively, if using normal vlans, this can also be achieved by enabling
access list on the switch and restrict traffic flows. Generally this is done
on a layer 3 switch, don't think it will support on your switch
On May 7, 2010, at 8:03, James Lamanna jlama...@gmail.com wrote:
On Thu, May 6, 2010 at 8:14 PM, Vineet Bhojnagarwala vbho...@gmail.com
wrote:
Alternatively, if using normal vlans, this can also be achieved by
enabling
access list on the switch and restrict traffic flows. Generally
Hi,
I've noticed that if a phone goes UNREACHABLE while it is Ringing,
when the phone comes back, Asterisk will not clear the channel that
was created, so it still thinks it is in the Ringing state.
The only way to clear this is to do a soft hangup on the SIP channel
or to restart Asterisk.
Hi,
I work for a small VoIP provider in Southern California. We are looking
for someone very knowledgeable in Asterisk to help work on the following:
- Maintenance of current Asterisk servers, updating Asterisk, monitoring
load, and other sysadmin tasks
- Devise and implement high-availability
It appears as though the 489 Bad Event response to the NAT keep alive
event responds to the local address, instead of responding to the
NATted address.
This causes Linksys phones to go amber (no registration) after a short
amount of time after placing calls.
Turning the Linksys NAT keep alive off
On Fri, Jun 18, 2010 at 10:51 PM, Stefan Schmidt s...@sil.at wrote:
James Lamanna schrieb:
It appears as though the 489 Bad Event response to the NAT keep alive
event responds to the local address, instead of responding to the
NATted address.
This causes Linksys phones to go amber
On Sun, Jun 20, 2010 at 5:42 AM, Ryan Wagoner rswago...@gmail.com wrote:
On Sat, Jun 19, 2010 at 12:00 PM, James Lamanna jlama...@gmail.com wrote:
On Fri, Jun 18, 2010 at 10:51 PM, Stefan Schmidt s...@sil.at wrote:
James Lamanna schrieb:
It appears as though the 489 Bad Event response
If you've used Linksys phones against recent Asterisk 1.4.x you may
have noticed
that they may drop registration for a quick bit and then go back to being ok
if your phone is behind NAT.
If you turn Asterisk's sip debug information on, you'll probably find
errors like these in your logs:
On Tue, Jun 22, 2010 at 12:06 PM, Stefan Schmidt s...@sil.at wrote:
James Lamanna schrieb:
If you've used Linksys phones against recent Asterisk 1.4.x you may
have noticed
that they may drop registration for a quick bit and then go back to being ok
if your phone is behind NAT.
If you turn
On Tue, Jun 22, 2010 at 4:31 PM, Ryan Wagoner rswago...@gmail.com wrote:
On Tue, Jun 22, 2010 at 6:26 PM, James Lamanna jlama...@gmail.com wrote:
On Tue, Jun 22, 2010 at 12:06 PM, Stefan Schmidt s...@sil.at wrote:
James Lamanna schrieb:
If you've used Linksys phones against recent Asterisk 1.4
On Tue, Jun 22, 2010 at 6:33 PM, Ryan Wagoner rswago...@gmail.com wrote:
--
The out of dialog support was the trick for 1.6.2.9 since it has
support for sending a keep-alive. I have attached a modified version
of your patch that worked for me. Do you mind if I attach the modified
version of
On Tue, Jun 22, 2010 at 8:57 PM, Andres and...@telesip.net wrote:
completely as well.
Below I've posted a patch that responds with a 200 OK to these
keep-alive requests, and I believe
also solves the temporary loss of registration problem, though more
testing in different environments
for
Hi,
Is there a way to poll the DND status of a Linksys SPA9xx/5xx phone?
The reason I ask is that I'm trying to implement DND + BLF on asterisk.
However, the DND softkey on the Linksys phone does not send any
feature codes to asterisk.
On the flip side, if you disable the Vertical Activation Codes
Hi,
I was wondering if anyone stumbled upon the correct event in a sip
NOTIFY (from a SUBSCRIBE)
to make the BLF lamps on a Linksys/Cisco SPA9xx/5xx go yellow?
I'm trying to differentiate between On the Phone and DND with the BLF.
Thanks.
-- James
--
Hi,
After upgrading to DAHDI 2.4.0 from Zaptel, I noticed a lot of HDLC
errors on my console:
[Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Bad
FCS (8) on Primary D-channel of span 1
[Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Abort
(6) on Primary D-channel
On Mon, Nov 29, 2010 at 10:02 AM, Shaun Ruffell sruff...@digium.com wrote:
On 11/27/2010 11:03 AM, James Lamanna wrote:
Hi,
After upgrading to DAHDI 2.4.0 from Zaptel, I noticed a lot of HDLC
errors on my console:
[Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Bad
FCS (8
See the following SIP trace.
Where in the world does Asterisk get port 1025 to respond to?
This is asterisk 1.6.x.
Thanks.
-- James
--- SIP read from zzz.zzz.zzz.44:9363 ---
NOTIFY sip:pbx1.mydomain.com SIP/2.0^M
Via: SIP/2.0/UDP 192.168.1.140:9363;branch=z9hG4bK-b9a860d3^M
From: xxx-xxx-
Hi Jeff,
On Thu, Jan 6, 2011 at 11:28 AM, Jeff LaCoursiere j...@sunfone.com wrote:
On Wed, 5 Jan 2011, James Lamanna wrote:
See the following SIP trace.
Where in the world does Asterisk get port 1025 to respond to?
This is asterisk 1.6.x.
Hi James,
I'm sure it would be the NAT
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