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Mindaugas Kuprys wrote:
> Hi,
> Can anyone suggest good quality IP phone with 2 Ethernet jacks. I wanted
> Sipura but they don't have such product.
Go for the Linksys SPA-942. It is what the Sipura SPA-841 evolved into.
- --
Ron W
FXO is coming from the PSTN, FXS is what devices connect to (like a analog phone).If you are using VOIP phone then you dont need the FXS modules, just FXO.On 8/24/06,
joea, j4computers <[EMAIL PROTECTED]> wrote:
As a complete newcomer to Asterisk, Digium and PBX, I have several questions.But I'll
spa3102router.html for the router
configuration as bridge and
http://www.wellsted.org.uk/spa3102voice.html for my voice configuration
with UK regionalisation (A-law, UK tones/cadences).
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No. 2021
or something else? Also then all the servers would have to be connected via NFS due to the fact of the voicemails wanting to be stored on another machine while its primary is down, or is this not even possible?
Any help would be great!
Thanks
Ron
help!
Ron
Bill Gibbs wrote:
I know but you could save some time and have it tested while
waiting...they might find a problem and save you a lot of headache. I
can tell you are one of the rare people who actually checks their stuff
before calling anyone but like another posted said, D Channels tend
ies.
Plus it
really helps to have all your ducks in a row when dealing with an ILEC - they
just don't seem to have much of a sense of humor about these things.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ron Gage
Sent: Thursday, August 17,
Quoting Bill Gibbs <[EMAIL PROTECTED]>:
What does the telco say when they test the circuit?
Bill
Bill:
I am having my "remote hands" check first on the Adtran that is feeding the
Asterisk box, then then go upstream from there.
Thanks for helping me see the obvious path to
te related) about "head of queue has not been transmitted
yet".
Ron
On 8/17/06, Ron Gage <[EMAIL PROTECTED]> wrote:
The PRI is connected at one end to an Adtran Atlas, I believe a 600.
The other end of the PRI is connected to a Digium T100. The two are
seperated
by roughly
yesterday and today I am getting D-Channel
errors.
Thanks for your assistance!
Ron
Quoting C F <[EMAIL PROTECTED]>:
What is the PRI connected to:
What hardware for the T1?
What Motherboard?
On 8/17/06, Ron Gage <[EMAIL PROTECTED]> wrote:
Hey guys:
I am having a bit of a problem
an Adtran Atlas.
HELP!
Ron Gage - Westland MI
[EMAIL PROTECTED]
This message was sent using IMP, the Internet Messaging Program.
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.conf
http://www.rongage.org/zapata.conf
Thanks for your help!
Ron Gage
Westland, MI
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y are going to get
harder to find.
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Ron Wellsted
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Hermann Wecke wrote:
> Ron Wellsted wrote:
>> I have been trying all the major distributors but they are all out of
>> stock with no dates for new stock to be delivered.
>
> As you are in the UK, why not talking directly to Bill
been discontinued?
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d the same problem until
upgrading to 1.2
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t; ++-+---+--+---+--+
> | 1 | default | 101 |1 | Macro | stdexten,101,sip/101 |
> | 2 | default | 102 |1 | Macro | stdexten,102,sip/102 |
>
> Thanks in advance for any help.
>
> */Jon Scottorn/*
> /Systems Admini
ny ideas on
> what could be causing SIP transfers to hang or drop?
>
> Thank you,
> Dan
Interestingly, we saw a very similar issue with 1.2.9.1 and Cisco 7960s
(SIP 8.2 fw) and HFC BRI ISDN cards last week, I went back to 1.2.7 and
every thing seems fine now.
- --
Ron Wellsted
[EM
fully investigate, I
reloaded 8.2 fw which registered straight away.
I did run a quick sip debug ip on the phone and saw asterisk replying to
the register request with 401 errors.
I hope to post more details when I get time.
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Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N
fully investigate, I
reloaded 8.2 fw which registered straight away.
I did run a quick sip debug ip on the phone and saw asterisk replying to
the register request with 401 errors.
I hope to post more details when I get time.
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N
recommend adding qualify=yes for all phones
behind NAT. Otherwise the router doing NAT may flush out the port
mappings relative to your phone. The qualify essentially sends a
keep-alive. We have Polycom IP500s and 501s and this works very well
for them (one sitting r
erisk ignore the IP in the SIP packets and look at
the TCP header. Also, qualify will send a 'keep-alive' to keep NAT
from losing the association of ports.
HTH,
-Ron
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is cdr
>
> (and that's all!)
>
>
> cdr_odbc.conf
> -
> [global]
> dsn=asterisk
does this dsn match up with the entry in your odbc.ini file and does
that use a driver that is in the odbcinst.ini file?
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wells
nfeld
Intel 537
Does your /etc/zaptel.conf file look like this:
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
loadzone=uk
defaultzone=uk
HTH
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Ron Wellsted
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N 52.567623, W 2.137621 Linux Counter No. 202120
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On Wednesday 22 March 2006 10:01, Nathan Alberti wrote:
>
> Telnet to the phone, login and type "show config"
Thanks Greg and Nathan!!
Ron
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On Wed, 22 Mar 2006, Simone Cittadini wrote:
Ron Wellsted ha scritto:
This is slowly driving me nuts!
I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk
1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing
On Wednesday 22 March 2006 00:33, Nathan Alberti wrote:
>
> Here is a dump of the configuration options, you will see there is a
> few new, these are also documented on the wiki.
>
Nathan,
How did you go about obtaining the dump ?
al restart).
Previously I was running 1.0.7 without this problem, I upgraded to fix a
problem with Monitor (the call stopped monitoring when transfered, 1.2.5
has fixed this).
Does any one have any suggestions?
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Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623,
Anyone have experience with the 3-08-2 release of Cisco's SIP firmware?
Are there any new features in the SIPDefault.cnf?
Thanks,
Ron
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list, I use the archives <http://lists.digium.com/mailman/listinfo/>
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;Dial"
softkey works perfectly wherever the call originated.
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SSL offloading, nothing like packet rewriting though.
So I think we are back to SER or a SBC from someone...
Thanks!
Ron
On 3/12/06, Gabriel Afana <[EMAIL PROTECTED]> wrote:
On a side note, the ServerIron can do Reverse-Nat
where it will rewrite the source IP to its Virtual IP an
e research will tell what ill be using in the
end :)
Once I get this going, I want to post a entire howto on the wiki.
Thanks!
RonOn 3/12/06, Gabriel Afana <[EMAIL PROTECTED]> wrote:
Hi Ron,
If the SBC would have served
mearly as a load balancer...I already have one and it didn&#
o frikkin docs for it!!!). We're using
OpenSER's send() command to forward registrations from a phone to all Asterisk
systems.
-----Original Message-From: Ron McCarthy
[mailto:[EMAIL PROTECTED]]Sent: Sunday, March 12, 2006 1:29
PMTo: Asterisk Users Mailing List - Non-Comm
from Juniper's site. Have you seen anything on this?
Thanks!
RonOn 3/11/06, Gabriel Afana <[EMAIL PROTECTED]> wrote:
Hi Ron,
I've been following your
thread. I noticed you mentioned about a Juniper Session Border
Controller. I checked online and read about it, but wa
Regarding OSPF, so your saying you have multiple * boxes setup with
same exact config and then just have OSPF fail everthing over to the
new server if it cant get to it? That makes sense, just never of even
thought of doing it that way. Heck, if you want to get real complex
just run BGP and you cou
Asterisk-Users] Clustering--Message: 6Date: Fri, 10 Mar 2006 12:22:12 -0700From: "Ron McCarthy" <
[EMAIL PROTECTED]>Subject: [Asterisk-Users] ClusteringTo: "Asterisk Users Mailing List - Non-Commercial Discussion"
? Ive been looking at using a
Juniper Session Border Controller, but not sure if thats gonna do the
trick, and then we also have SER..
Any comments would be great!
Thanks
Ron
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I havent got any mails since 2:42 this morning..usually i get at least
the normal 10-15 a hour, if someone gets this can they reply?
Thanks!
Ron
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>
The SIP firmware does not allow the softkeys to be programmed :(
Unfortunately you have to make a choice:
SIP firmware - Easy to implement on *, but poor XML support
SCCP firmware - poor/non-trivial asterisk support, great XML support.
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.we
Also, SATA on a onboard SATA card will eat more CPU then a SCSI system.
Are you running software RAID by chance with your SATA? SCSI or SCSI
Raid will not each CPU near as much since the HBA does all the work and
does tie up the CPU with all its I/O's. We have successfulyl recorded
5+ calls at a ti
er=yes
dropcount=2
maxjitterbuffer=500
maxexcessbuffer=300
minexcessbuffer=60
jittershrinkrate=1
maxjitterinterps=10
resyncthreshold=1500
Thanks,
-Ron
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somemore then, I figure it would be playing a part into this!
Thanks for the help!
RonOn 2/25/06, yusuf <[EMAIL PROTECTED]> wrote:
Ron McCarthy wrote:> Hi List,>> Im planning on setting up asterisk for a large scale enviorment, with> multiple sites.>> We will be doing qui
ad of having it go
to the data center.
My main concern is the dialplan, I guess if the peer is not local it
would then go out the IAX or SIP gateway to the main * server and then
check in its dial plan/routing table there, correct?
Any help/suggesstion on this would be great!
Thank
.
> exten => 850,1,Goto(Mercury-Network,850,1)
> exten => 888,1,VoiceMailMain(@Mercury-Network-Emp)
Try adding
dtmfmode=rfc2833
to your sip entry.
Also, check the permissions for the file on your boot server.
HTH,
-Ron
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I would like to kick off an AGI script when an agent answers a call...
thus passing the phone that answered the call, the CID, etc.
Anyone know how I could do this?
TIA
-Ron
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Hello all,
Has anyone figured out a way to send email notifications etc. due to
failed IAX2 registration attempts?
Thanks
-Ron
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Thanks for all your responses. The reason we would not go through a
provider is that I run Asterisk phone systems, we have access to
bandwidth, and I can do this myself for a fraction of the cost.
Cheers
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> I just did a quick office poll and everyone agreed if a party candidate
> did this to them, they would vote for the candidate's opponent. The office
> is rarely unanimous in political matters so this was a pretty interesting
> result to me.
>
> I'm pretty sure the feeling is universal.
>
> Like I
> Ron Senykoff <[EMAIL PROTECTED]> wrote:
> > I'm helping out with a political campaign and would like to use asterisk
> > to blast out about 200,000 calls with a short message from the candidate.
>
> Can you tell me which party this is for, so I can ensure I neve
eate call files. Use asterisk manager interface to
monitor calls and that way I can keep the preset concurrent limit.
Any ideas?
TIA!
-Ron
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. are you routing your phone calls via VoIP? If so, it is not
recommended to run FAX via VoIP. The two don't mix. FAX is not able to
handle packet loss like VoIP. Also, any codec other than uLaw will not
even come close to working, as the codecs are designed to comp
> It does take between 1 and 12 hours for the new settings to take effect.
> Dan
I've recently had a problem with codec changes taking affect, but they
were nice enough to on-the-fly move an 800 number to route from one
site to another. It seems there was some kind of cacheing issue, as I
changed
like you describe. Provided the topology of your switches is OK, you
should be fine. Just don't uplink to another switch where you can
create a non-QoSd bottleneck link.
-Ron
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As
5.1 SIP firmware, it should contain:
P0S3-05-1-00
P0S3-05-1-00
Ideally, you should be looking to run at least V6.3 SIP firmware, V7.4
or 7.5 would be much better.
HTH
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No. 2021
is that the new
sip.cfg now contains ntp settings. You'll need to modify these to fit
your timeserver setup.
-Ron
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> One thing I was pondering: you are not, by chance, using the same
> sip.cfg between version 1.4.1 and version 1.6.2 are you? The file has
> changed significantly between these versions, and certain acoustic
> settings that worked with 1.4.1 may not work with 1.6.2 (Not to mention
> that ipmid.cf
an inaccurate match?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ron hotmailSent: Friday, January 27, 2006 8:17
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] OT?:
International number parsing
I
It realy is a pain in the *ss.
the problem is just how you explained. when
trying to match the terminating number, there's no SINGLE fixed pattern for the
dialcodes. so how do you know how many digits of the term number to match
against the dialcode? you dont. you have to match the dialcod
> I've been running 1.6.4.0064 for the last few weeks..
> I've had no problems with it, I haven't done a whole lot of speaker
> phone with it yet though.. Once my IP4000 reboots It'll be running it as
> well so that will be a good test.
Which boot
> > I'm running 1.6.2.0041 according to my phone.
> >
> > Which firmware "worked" for you?
>
> It was the old firmware from when we first got the phones actually.
> 1.4.x I think. Then I read that they fixed the CID issue and decided
> we needed an upgrade. I tried it out on my phone, but didn't re
> I'm running 1.6.2.0041 according to my phone.
>
> Which firmware "worked" for you?
It was the old firmware from when we first got the phones actually.
1.4.x I think. Then I read that they fixed the CID issue and decided
we needed an upgrade. I tried it out on my phone, but didn't really
notice t
s with 3 separate
Asterisk servers. And I have to reiterate... all was good until the
firmware upgrade.
-Ron
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place a call to my inbound number, the call comes into asterisk, runs
through the steps, then comes in again, and runs through the steps
again. On the calling phone there is silence until the second call
finishes, then I get busy/congestion.
TIA
- --
Ron Wellsted
[EMAIL PROTECTED] http
ng now however.
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Ron Wellsted
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en able to find is the rez of the 7940g : 100x145.
>
> -Dan
The usable resolution for the programmable (logo url) bitmap is 90 x 56.
From:
http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx
- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623,
hmm. That's a tad awkward
:-|
- Original Message -
From: "Douglas Garstang" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, January 19, 2006 5:00 PM
Subject: RE: [Asterisk-Users] Dundi Examples
The O'Reilly TFOT book is full of
#x27;s why it's asking for 7960
> files when it's a 7940. Or perhaps there is an error in my SEP
> file? Here's the contents of that file:
- - - - 8<
> Kris
Your existing SEPmacaddress.xml file has far too much in it. In order
to upgrade to SIP, just use
P0S3-07-5-00
, other Finarea sites (1899.com, 18185.co.uk) do use https.
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Ron Wellsted
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(happen to be booted to the windows side at
the moment) and for some reason Firefox is working fine now.
It was always set to accept cookies. If I get the error again, I will
repost.
Thanks again.
Ron
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You were right on. I turned down my security settings in IE and it went
into the config just fine. It's the simple things...
I found the Firefox issue earlier as well.
Thanks for the help, guys.
Ron
Philip Edelbrock wrote:
Ron Bulthuis wrote:
I just purchased a Grandstream gxp
using default firmware
of 1.0.1.9 but it shouldn't be even looking at Asterisk yet.
Do I need something more just to browse to these configuration pages in
the device? All 3 units are doing the same thing.
(I did not find anything in the FAQ's or docu
or
> now it works.
> I feel sure the rules and results are different in every jurisdiction.
> Is LNP even allowed in the UK or the EU?
>
Within the UK, Number Portability between providers of the same type of
service is a legal requirement. Since we charge differently for calls
on l
phone?
> ( Like in Xlite we do: "System Settings"-> "Network" -> "Listen RTP port" )
>
> Thanks
> Joao Pereira
AFAIK, the firmware requires the RTP port range to be above 16384. My
7960 with 7.5 SIP firmware is working fine on 18060 - 18078
- --
Ron W
,
Ron Arts
NeoNova.nl
Adam Robins wrote:
We have built an Asterisk network using an MPLS-based IP VPN. We have
one location in New Brunswick Canada that consistently gives us major
quality problems, whereas the others are flawless. Quality problems
take the form of static, poor voice tonality
ag or
>>>something to tell Asterisk not to install sound files, or at the very
>>>least not to overwrite ones already existing?
Why not add a separate "make sounds-install" or "make sounds-upgrade"
step post-install.
It will never be possible to come up with a
gt; Help is always appreciated! J
>
>
>
> Thank you,
>
> Scott Miller
How about the Cut application (depreciated)
<http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+cut>
or the CUT function
<http://www.voip-info.org/tiki-index.php?page=Asterisk+func+cut>
hrough my SPA3K using 711u) ...
Thanks a lot ;)
Julien.
Is the 794G configured to use a VLAN ? I had a similar problem with VLAN
and ARP packets. I got around this by not using VLANS but putting the
phone on a separate subnet (with QoS on the phone subnet).
- --
Ron Wellsted
http://www.
lar TFTP
>> daemons? I cannot seem to get mine to speak to a Linux box, but
>> Solarwinds under Windows works like a charm.
>>
Some versions of Cisco firmware require the tftp server to be specified
with option 150, not option 66 (tftp server name)
- --
Ron Wellsted
http://www.
from the switches.
How often are you rebooting the phones? The last time I had to reboot
the phones was to upgrade to SIP 7.5
- --
Ron Wellsted
http://www.wellsted.org.uk
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The Netgear FSM7326P switch also supports the Cisco Pre-Standard directly.
We have these powering all our CP7960 phones perfectly.
- --
Ron Wellsted
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und to set this in
tftp-config file etc.
Thanks in advance,
Roland
The timeout is set in the dialplan.xml file with the Timeout tag. Like
this:
HTH
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Ron Wellsted
http://www.wellsted.org.uk
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tively new to Gentoo
> so I don't have much experience to tell me if it's a distribution issue
> or an asterisk/zaptel issue.
>
> Thanks,
> Waldo
Also running Asterisk 1.0.7, currently on kernel 2.6.11-Gentoo-r6
The dmesg output suggests that zaptel wants the crc-ccitt mo
inbound
driver=i4l
language=en
type=autodetect
stripmsd=0
dialtype=tone
mode=immediate
msn=yourMSNhere
group=9
dtmfmode=asterisk
incomingmsn=*
device => /dev/ttyI0
device => /dev/ttyI1
You _WILL_ need to set your MSN (change yourMSNhere to your full MSN,
usually without the area co
.
Any help on obtaining the updated firmware quickly and painlessly would be
great... :-)
Cheers
M
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Ron Wellsted
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> Hi Michael,
> d
ten => _9.,3,Dial(${VOIP1}...)
exten => _9.,4,Hangup()
exten => _9.,103,NoOp()
exten => _9.,104,SetGroup(voip2)
exten => _9.,105,Dial(${VOIP2}...)
exten => _9.,106,Hangup()
exten => _9.,205,NoOp()
exten => _9.,206,Playtones(congestion)
exten => _9.,207,Congestion()
exte
both the
> console attachment as well as the background process. I need to attach
> to the running asterisk in order to do "init keys" but once I do that,
> it seems I cannot just let it go into the background again.
>
> Any suggestions most welcome.
>
try "q
ny idea ?
>
Try
[SIP-OUT]
type=peer
host=10.XX.XX.XX
port=5061
defaultip=10.XX.XX.XX
HTH
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961 Gossiptel:9309811
N 52.567623, W 2.137621
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learn/is a friend/relative of a PBX supplier who is being usurped? Have
you made an enemy of this person?
We have just switched over to Asterisk with 7960s. We have had a few
little problems but have not lost a call yet. OK, we have left a few
callers on hold a bit longer than we intended, once or tw
sound
files.
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD: 519961
N 52.567623, W 2.137621
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for paging, etc.
>
>
>
> Please don?t flame me. Just getting into PBX?s and haven?t had much
> experience with them.
2.) Registering the phone with more than one server (possibly in
different parts of the world).
3.) Different caller IDs
Personally, I use the e
functionality and build quality.
They are also the best speaker phone for small conferences.
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961 Gossiptel:9309811
N 52.567623, W 2.137621
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go, or must it
> tftp from the head server?
>
> Regards,
> Greg
You could try CDW or Insight, see <http://www.voip-info.org/wiki-Cisco>
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961 Gossiptel:9309811
N 52.567623, W 2.137621
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Simon Morris wrote:
> On Fri, 2005-04-22 at 11:13 +0100, Simon Morris wrote:
>
>>On Thu, 2005-04-21 at 21:36 +0100, Ron Wellsted wrote:
>>
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>&g
risk stops in
> the first playing. Someone have same problem or can help me?
>
Have you compiled and installed zaptel and loaded ztdummy?
HTH
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961 Gossiptel:9309811
N 52.567623, W 2.137621
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then lines 5 and 6 appear on the phone console.
Weird!
/rg
There is a known issue with the size of the SIP*.cnf files on some
versions of the firmware. The cure for this is to stripout all comments
from the .cnf files.
HTH
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD: 519
p()
then program the messages button to dial 8501 either via settings, SIP
Settings, Messages URI (set to 8501) or in the SIPDefault.cnf file
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961 Gossiptel:9309811
N 52.567623, W 2.137621
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nyone else found themselves in a similar situation? If so, how did
you work around it?
Is it worth raising a TAC with Cisco?
- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961 Gossiptel:9309811
N 52.567623, W 2.137621
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