On Fri, 2019-10-11 at 14:12 -0400, Brian J. Murrell wrote:
> I'm trying to clarify my understand of gosub, macros and AEL.
>
> My understanding is that macros using the Macro() application, which
> is
> defined in extensions.conf by:
>
> [macro-foo]
> ...
>
>
I'm trying to clarify my understand of gosub, macros and AEL.
My understanding is that macros using the Macro() application, which is
defined in extensions.conf by:
[macro-foo]
...
and called in extensions.conf with
exten => _9NXXNXX.,n,Macro(fastbusy)
is deprecated in favour of Gosub().
Using Asteirsk 13.28.1:
If I configure my pjsip transport to handle NAT from the Internet:
[transport-tcp]
type=transport
protocol=tcp
bind=10.75.22.8:5060
local_net=10.75.22.0/24
external_media_address=[external address redacted]
external_signaling_address=[external address redacted]
When a
On Fri, 2019-09-13 at 14:21 +0200, Administrator TOOTAI wrote:
>
> Escape it with \
Tried that. It doesn't work.
Cheers,
b.
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How can I use an IF statement with a true value being a variable that
has a colon in it? The colon in the true value variable is being taken
as the delimiter for the false value.
The only solution I came up with was some hackery to use STRREPLACE to
replace the : with a % before the IF statement
On Tue, 2019-07-16 at 16:20 -0400, Brian J. Murrell wrote:
> Is there any option to prevent Asterisk from rewriting the To:
> address
> of a SIP MESSAGE that it's received and will forward to another SIP
> client?
>
> That is, when Asterisk receives a MESSAGE with the
Is there any option to prevent Asterisk from rewriting the To: address
of a SIP MESSAGE that it's received and will forward to another SIP
client?
That is, when Asterisk receives a MESSAGE with the To; header saying:
To:
and wants to forward that to foo@10.75.22.100, I'd like the To: header
to
On Thu, 2019-06-06 at 09:33 -0400, Brian J. Murrell wrote:
> I'm trying to use linphone-android with asterisk but there is an
> aspect
> of the way asterisk and linphone-android interact with MESSAGE
> transactions that is causing problems.
>
> The linphone-android f
I'm trying to use linphone-android with asterisk but there is an aspect
of the way asterisk and linphone-android interact with MESSAGE
transactions that is causing problems.
The linphone-android folks consider both the To: and From: address in
MESSAGE transactions when deciding which "chat" to
On Wed, 2019-04-17 at 13:50 -0400, Joshua C. Colp wrote:
>
> The same escaping should apply there for extensions.conf as it's a
> config file thing, I don't use AEL and don't know anything in that
> regard. It may work the same way there.
How very odd. It is working now. I am sure I did
On Wed, 2019-04-17 at 11:56 -0400, Joshua C. Colp wrote:
> On Wed, Apr 17, 2019, at 12:51 PM, Brian J. Murrell wrote:
> >
> > I can add it onto the end of the variable in the Dial() command:
> >
> > Dial(${FRED};transport=tcp,${timeout},TtWw);
[ the part you trimmed
On Wed, 2019-04-17 at 10:04 -0400, Joshua C. Colp wrote:
>
> You specify the transport in the SIP URI. For example:
>
> sip:t...@example.com;transport=tcp
Hrm. This is probably going to be pretty basic, but some googling
didn't seem to come up with anything. How do you do this when you are
Hi,
I'm using Asterisk 13.x and have defined a pjsip TCP IPv6 transport:
[transport-tcp-ipv6]
type=transport
protocol=tcp
bind=[2001:1234:5678:abcd::2]:5060
I also have an IPv4 version of that:
[transport-tcp-ipv4]
type=transport
protocol=tcp
bind=10.75.22.8:5060
I've then configured an
On Thu, 2019-04-04 at 15:08 +0200, Antony Stone wrote:
>
> It's not "Password", it's "Secret" :)
Ha ha. I knew it would be a head-smack type problem.
Cheers,
b.
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I'm not sure how much more simple I can make this but I just cannot
seem to get my Asterisk 13 to accept a connection on the manager
interface:
--- manager.conf ---
[general]
enabled = yes
port = 5038
bindaddr = 127.0.0.1
[myasterisk]
secret=a
permit=0.0.0.0/0.0.0.0
read = all
write = all
So,
On Thu, 2019-02-21 at 11:17 -0500, Brian J. Murrell wrote:
> In the past, I have created variables that hold multiple extensions
> such as:
>
> HOUSEPHONES=PJSIP/mom/grandma
>
> so that I can do a Dial(${HOUSEPHONES},...) with it, to ring multiple
> phones.
>
> B
On Fri, 2019-03-01 at 15:54 -0500, Joshua C. Colp wrote:
>
> That's correct. You'd either need to retrieve the line parameter from
> the outbound registration or forge the source IP address,
Can I eliminate the identify by IP address then, given that my ITSP is
supporting the line parameter? Or
On Fri, 2019-03-01 at 15:41 -0500, Joshua C. Colp wrote:
>
> I don't understand what you mean. Your ITSP has stated that they
> don't want you to do authentication with them, so you can't.
They are implying, as I am understanding them, that somehow SIP packets
they send me shouldn't need to be
On Fri, 2019-03-01 at 14:15 -0500, Joshua C. Colp wrote:
> you can try line functionality on the outbound registration which
> may or may not work[2] (requires the upstream to adhere to the RFC,
> which not all do).
My provider seems to implement this.
However even with the line=... in the:
SIP
On Fri, 2019-03-01 at 14:15 -0500, Joshua C. Colp wrote:
>
> You either configure IP based matching using an identify section[1]
That's what I did:
[itsp]
type=registration
transport=transport-udp
outbound_auth=itsp-auth
server_uri=sip:pop1.itsp.example.com
I'm being told by my ITSP that my Asterisk shouldn't be challenging
their system to authenticate (i.e. a 401 response) when they send me a
SIP MESSAGE (or I suppose a SIP INVITE for that matter).
But I'm not sure what a pjsip.conf configuration for that looks like.
How does one associate an
On Sun, 2019-02-17 at 17:31 -0500, Brian J. Murrell wrote:
> I have a PJSIP trunk set up which works fine for voice. I can call
> out
> and I receive calls from it once it registers.
>
> What isn't working though is receiving MESSAGE (i.e. SIP SIMPLE)
> events. It was wor
In the past, I have created variables that hold multiple extensions
such as:
HOUSEPHONES=PJSIP/mom/grandma
so that I can do a Dial(${HOUSEPHONES},...) with it, to ring multiple
phones.
But now some of those phones will be registering multiple times and
thus have multiple contacts, so I want to
On Wed, 2019-02-20 at 12:38 -0700, John Kiniston wrote:
> I don't see any other messages in this thread other than your initial
> one
> and my response, perhaps the listserv hasn't relayed it to me yet.
I started a new thread:
On Wed, 2019-02-20 at 11:46 -0700, John Kiniston wrote:
> Use the IF function to evaluate and change the dial command directly.
Thanks for taking the time, but that doesn't actually answer the
question I asked. It in fact answers the caveat I specifically
mentioned:
> Granted the particular
Following up on my previously asked question if I rewrite the branching
example (not that it negates the more general branching question) I was
using as such:
exten =>
s,n,Set(EXT=${IF($[${SIP}=PJSIP]?${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,PJSIP/,)})}:${ARG2})})
exten =>
Is there any less cumbersome way of doing conditionalized/branching in
extensions.conf other than something like:
exten => s,n,GotoIf($["${SIP}" = "PJSIP" ]?pjsip)
exten => s,n,Dial(${ARG2},20,TtWw)
exten => s,n,Goto(afterdial)
exten =>
I have a PJSIP trunk set up which works fine for voice. I can call out
and I receive calls from it once it registers.
What isn't working though is receiving MESSAGE (i.e. SIP SIMPLE)
events. It was working earlier today but I seem to have done something
as I was enabling voice on the trunk to
On Mon, 2019-01-28 at 07:29 -0700, George Joseph wrote:
> When you have an "identify" object configured, you should just use
> "ip" as
> the "identify_by",
But isn't ip the highest priory check in the default value of
endpoint_identifier_order and by extension, wouldn't an endpoint
without an
On Mon, 2019-01-28 at 07:29 -0700, George Joseph wrote:
>
> What version of Asterisk
13.11.1
I know, I could stand to upgrade.
> and what's the value of the "identify_by"
> parameter for the endpoint?
It doesn't have one. I guess you are implying it should have one.
> When you have an
I have a trunk set up for the DID from my provider:
[my_provider]
type=registration
outbound_auth=my_provider
server_uri=sip:sip.example.com
client_uri=sip:my_usern...@sip.example.com
retry_interval=60
[my_provider]
type=auth
auth_type=userpass
password=123456
username=my_username
On Tue, 2019-01-15 at 12:01 -0500, Joshua C. Colp wrote:
>
> The chan_sip module has this implemented under the "nat" option using
> "comedia" as I recall.
Yeah. The help for which reads:
Send media to the port Asterisk received it from regardless
of where the SDP says to send it.
> It causes
On Tue, 2019-01-15 at 09:00 -0700, John Kiniston wrote:
> How is your endpoint currently configured in asterisk?
It's configured as a chan_sip peer.
> Have you tried
> rtp_symmetric to see if the endpoint sends audio to asterisk if
> asterisk
> can send audio back to the client?
That would
This is going to be a bit of an odd situation, but perhaps might become
more and more common (as mobile phone SIP clients utilize PUSH proxies
instead of the battery draining direct registering with SIP servers).
I have a SIP client which can be on the same RFC-1918 LAN as my
Asterisk server.
Hi,
I want to be able to send SIP SIMPLE messages via/to my VOIP provider
but in trying to do so with MessageSend() I am getting 401 errors back
from them, unsurprisingly. They want such messages from me
authenticated with my account just as they would for SIP voice calls.
For voice calls, of
I am trying to use the FILTER() function to strip out / from a CID
name. I have the following in my extensions.conf where I want to
perform the filtering:
exten = s,n,Set(NAME=${FILTER(\x20-\x2e\x30-\7d,${DIAL_NAME})})
However, when ${DIAL_NAME} is, say, J J DOE the string resulting
from the
I want to use Asterisk as a general message delivery system here.
That is, I want to be able to have a (shell, perl, etc.) script on my
Asterisk server dial an extension, wait for it to be answered and then
play a sound file and then hang up, or even wait for a response or
reactions to some IVR.
I wonder if Asterisk's skinny/sccp channel driver could be used as a
client to register with a Cisco PBX. That is, along with a SIP
client, say, have Asterisk and said SIP client stand in for a Cisco
phone, or an IP Communicator.
Anyone done this?
Cheers,
b.
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On Wed, 2010-03-17 at 10:56 -0500, Jason Parker wrote:
No, this isn't currently possible.
Damn.
I did ponder this for a while, but my
conclusion was that the effort required to do so would far outweigh any
benefit
you'd gain from it.
How about having something -- anything on Linux,
On Wed, 21 Jan 2009 19:02:01 -0500, Steve Totaro wrote:
Why not just get a softphone and use a USB soundcard or even the onboard
sound card as your ATA? Like a MagicJack and SJphone or Xlite or
whatever it is that works with it.
Please forgive my ignorance but I am not following you.
My
Slightly OT, but I'm wondering if anyone here has come across a soft
ATA. That is, software that will perform the functions of a basic POTS
line ATA on Linux with a zaptel driven card.
I have a Linux machine with a zaptel card in it and I want to have
another Linux machine running Asterisk
So I have (and have had) jabber configured for some time, specifically
for GTalk, but something has occurred to me. If somebody happens to
send an IM (text) to that account, nobody is going to be receiving it.
I'd like to send a canned message back to any sender of an IM.
Possible?
b.
On Mon, 2008-10-27 at 10:11 -0400, Brendan Martens wrote:
I found this in the sample voicemail.conf:
;pollmailboxes=no; If mailboxes are changed anywhere outside of
app_voicemail,
;; then this option must be enabled for MWI to
work. This
;
On Mon, 2008-10-27 at 11:13 -0500, Mark Michelson wrote:
The behavior you are seeing is most likely due to SIP's MWI behavior in
Asterisk
1.4. The way it works is to poll the mailboxes every so often to see if new
messages are available.
Yeah, that sounds like it.
2. If you want to
I've configured asterisk 1.4 to use imap storage for voice-mail and
while I'm happy with it generally speaking it really seem to hammer the
IMAP server. It appear, from the IMAP server logs that it's polling
the imap server every *second* for mailbox updates for the users'
voice-mail folders.
On Sun, 2008-10-19 at 13:31 +0300, Kevin P. Fleming wrote:
Asterisk 1.6 supports proper SRV record sorting, so that the lookups
will return the correct record to the module that requested the lookup.
That would be in ast_get_srv() then? In 1.4.17 I do see weight
processing but not priority
On Sun, 2008-10-19 at 21:14 +0300, Kevin P. Fleming wrote:
Never mind... I was mistaken. The srv_callback() function puts the
records returned by the DNS lookup into priority order (lowest numbers
first),
Yes, I can see that function and while I have not audited it detail, it
looks as though
Given the following SRV records:
_sip._udp.tollfree.sip-happens.com. 38400 IN SRV 10 0 5060
sometimes.sip-happens.com.
_sip._udp.tollfree.sip-happens.com. 38400 IN SRV 20 0 5070 ares.sip-happens.com.
Why is asterisk (1.4.17) not honouring the priority and not failing over
to using other records
On Fri, 2008-10-17 at 10:32 -0500, Andres wrote:
Because Asterisk does not support that.
Which is just another way of saying Asterisk is broken then. SRV
records have requirements for their correct use. If those requirements
are ignored, that is a broken implementation.
The only thing that
On Fri, 2008-10-17 at 11:18 -0500, Eric ManxPower Wieling wrote:
It should be fairly easy to write an AGI script that does the SRV query,
do whatever you want with the response, set a channel variable with
the results and use that in your dialplan.
Maybe. If I were an AGI hacker. But
On Fri, 2008-10-17 at 11:35 -0500, Eric ManxPower Wieling wrote:
If you fight Asterisk's oddities then you will have a depressing and
miserable life. If you embrace Asterisk's oddities then you will have a
joyous and enlightened life. 8-)
I just want something that works. :-)
I agree
On Fri, 2008-10-17 at 12:11 -0500, Tilghman Lesher wrote:
Have you considered upgrading to 1.6?
Not to this point, no. 1.4 does everything I want and if it ain't
broke, don't fix it. Well, now it's broke I guess. Still, Ubuntu still
uses 1.4 and I don't like having to maintain my own
I'm looking at the app_voicemail.c from both 1.4 and 1.6.1 and seeing
that neither allows an individual mailbox to override the imapfolder
value. It seems entirely intuitive to me that one might want to do
that, not to mention how trivial it looks to add that to
app_voicemail.c.
Maybe my
On Tue, 2008-09-30 at 08:23 -0500, Lyle Giese wrote:
1) a two line phone can register with two different * servers or sip
carriers.
Indeed. But if I only had the one * server which itself registered to
my carriers...
2) It's easy for both incoming and outgoing to separate business from
On Tue, 2008-09-30 at 17:29 -0500, Lyle Giese wrote:
I have never been convinced that VM via email is a convenence. You
have to use the loudspeakers on the PC or headphones, which is not as
convenient as a handset.
Depends on your working environment I guess.
Not to mention the privacy
I'm looking into getting a new phone and wondering what the difference
in functionality is between a single line phone with call waiting and a
real 2 line phone (either a real SIP phone or an analog 2 line phone and
a 2 port ATA) is. Why would I want the real 2 lines vs. just being able
to take
On Sat, 2008-09-27 at 15:11 -0400, Dean Collins wrote:
Who is Chris Langford in Huntsville Alabama and is he seeking Digium’s
permission in order to report the asterisk mailing lists out onto the
internet
Digium's permission? Does Digium own the copyright on what I write? I
think not. I do.
On Fri, 2008-09-26 at 08:43 +0100, Grey Man wrote:
It's not particularly difficult to determine the best IP address for a
piece of client software to use.
Oh?
Check the local machines default
gateway, apply the subnet mask and then compare it against all the
local IP's.
Yeah? And if
On Fri, 2008-09-26 at 14:54 +0100, Grey Man wrote:
On Fri, Sep 26, 2008 at 2:36 PM, Brian J. Murrell [EMAIL PROTECTED] wrote:
Yeah? And if more than one matches? Then what?
Use one of them!
And if the one I choose to use doesn't work because of some kind of
policy routing or filtering
On Fri, 2008-09-26 at 10:16 -0400, SIP wrote:
The RFCs are there for a reason. All SIP forking is UAS territory. Not
UAC territory.
From http://bugzilla.gnome.org/show_bug.cgi?id=553810 Damien Sandras
asks:
I repeat, Ekiga is doing something perfectly legal.
The real
On Fri, 2008-09-26 at 10:41 -0400, SIP wrote:
Oh yes. It's perfectly legal.
It's also a) NOT SIP forking, b) Lazy, and c) Poorly designed.
Sending multiple requests and hoping and praying that the recipient will
ignore two of them (it will NOT in many cases -- specifically set out by
the
On Thu, 2008-09-25 at 20:49 +0300, Tzafrir Cohen wrote:
Great news! You mean that there is finally a free implementation of the
skype protocol so I can start using it?
Free? AFAICT, not. Neither free as in beer nor speech. Move along,
nothing to see here.
b.
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So, I have been testing ekiga 3.0 with Asterisk, and sadly, it don't
work. I am told by the ekiga devs in
http://bugzilla.gnome.org/show_bug.cgi?id=553595 and
http://bugzilla.gnome.org/show_bug.cgi?id=553810 that the problem is
that Asterisk does not support SIP forking.
The issue is that I have
On Thu, 2008-09-25 at 14:56 -0400, SIP wrote:
Sending from multiple different points of origin doesn't make any sense
at all in either a logical or rational fashion. What's it supposed to
accomplish?
It seems to be a shot-gun approach to making a SIP connection. The
assumption being I suppose
On Thu, 2008-09-25 at 15:31 -0400, SIP wrote:
That strikes me as being careless and unreliable.
That's one argument. I can also see the ekiga developers' argument
though and that's to strive for the most automatic functionality
possible. The less things you have to ask users, the more likely
On Thu, 2008-09-25 at 17:25 -0700, Fred Posner wrote:
I talked with both Skype and Digium today at Astricon for a while on this...
it's actually going to be amazing.
It's still early, but still, nobody has answered my question as to
whether Skype will be using my Asterisk server's CPU and
On Thu, 2008-08-14 at 14:15 +0200, Klaus Darilion wrote:
Use the ENUMLOOKUP function, e.g.:
And take note that it's very naive. See my previous posting for an enum
AGI that is more intelligent. The only thing it does not do that I
would like to add is give up on the DNS lookup much earlier
On Wed, 2008-07-16 at 15:08 +1000, Rob Hillis wrote:
I hadn't realised this was for a home server... yes I agree, for a home
server the Digium or Sangoma cards are a little too expensive.
Indeed.
I can't speak for the SPA-3102, but the SPA-3000 I use here at home
doesn't do a brilliant
On Wed, 2008-07-16 at 09:31 -0400, Brian J. Murrell wrote:
IIRC -- this was 7+ years ago
Er, I mean 2+ years ago, just to keep the facts straight.
b.
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On Sun, 2008-07-13 at 10:22 -0400, Brian J. Murrell wrote:
I have a wildcard 100 xp on my pots line and all was working just fine
up until a few days ago when all of a sudden it stopped receiving caller
id on incoming calls. I know caller id is being presented on the line
as the analog set
On Tue, 2008-07-15 at 22:31 +1000, Rob Hillis wrote:
Brian J. Murrell wrote:
One thing I have noticed is that in the cases where the wildcard cannot
determine the CID (i.e. because the rxgain is up around 10.5), I get
this in my asterisk console:
[Jul 15 08:04:09] NOTICE[26696
On Tue, 2008-07-15 at 12:49 -0400, Noah Miller wrote:
It is odd that it would work one day and not the next.
Indeed.
I'd have to
say, though that I've seen that rxgain/txgain values beyond +-8 seem
to yield unpredictable results in many areas,
Yeah, I was pretty alarmed months ago when I
I have a wildcard 100 xp on my pots line and all was working just fine
up until a few days ago when all of a sudden it stopped receiving caller
id on incoming calls. I know caller id is being presented on the line
as the analog set on the same line always gets it.
What is strange is that this
On Mon, 2008-06-30 at 12:03 -0400, Andrew Joakimsen wrote:
Does anyone know of a spam filter that will work with Asterisk?
What does spam have to do with Asterisk? Or do you mean spit perhaps?
http://en.wikipedia.org/wiki/VoIP_spam ? Probably the same techniques
such as whilelisting,
On Mon, 2008-06-30 at 11:15 -0500, spectro wrote:
I need a way to block that IP from connecting to my
asterisk server, please advice.
netfilter. aka iptables.
b.
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I have an outbound-ld context as follows:
[ Context 'outbound-ld' created by 'pbx_config' ]
'_1NXXNXX' = 1. Macro(enumdial|${EXTEN}) [pbx_config]
102. Wait(1) [pbx_config]
103.
On Wed, 2008-06-25 at 11:25 -0500, Tilghman Lesher wrote:
That's only true within the same context. ONLY if a match is not found in the
current context will it go into an included context.
Ahhh. Well, then that explains it. Any thoughts on how to achieve my
goal, without having to encode all
On Sun, 2008-06-15 at 17:43 +0300, James Mutuku wrote:
Please advice on channel bank
Dude. There's the cool new website you should check out. It's
www.google.com.
Seriously. This list is not full of people waiting to do the simplest
research at your request. Spend a few minutes and do some
On Sun, 2008-06-15 at 11:03 -0400, Steve Totaro wrote:
On Sun, Jun 15, 2008 at 10:53 AM, Brian J. Murrell
[EMAIL PROTECTED] wrote:
On Sun, 2008-06-15 at 17:43 +0300, James Mutuku wrote:
Please advice on channel bank
Dude. There's the cool new website you should check out. It's
Right now I have an Asterisk 1.4.18ish server and a Wildcard POTS
interface. As it is now, when the zap line gets a call, Asterisk
answers it and waits for the analog CID to be presented, then rings the
SIP phones with the call and the CID. There's a significant latency
involved in doing this.
On Wed, 2008-06-11 at 10:12 -0400, Steve Totaro wrote:
Do you actually have callerID on your line? That takes about two
seconds. Try removing it and see how much faster Asterisk answers.
That brings up a question though, on a regular landline with caller ID
the phone rings right away, it
On Wed, 2008-06-11 at 10:38 -0400, Steve Totaro wrote:
Exactly! It is funny how when idea or technology is ready, many
people have the same thougts at the same time.
Indeed. But what is even more interesting is that this technology is
not just ready. It's been ready for a long time and
On Wed, 2008-06-11 at 15:57 +0100, Gordon Henderson wrote:
Intersting idea... However, I live in a country where on a regular
landline with caller ID, the caller ID is displayed before the phone
rings, so make sure it's an option and not hard-wired...
Well, I think your situation makes the
On Wed, 2008-06-11 at 13:30 -0500, Brent Davidson wrote:
On the subject of CallerID and ringing, I'm not sure if it's like this
everywhere in the US, but where I live in Texas, our caller ID signal
is sent between the first and second rings.
It's like that here in Canada too.
If the phone is
On Wed, 2008-06-11 at 14:51 -0400, Steve Totaro wrote:
If you ever have problems with a call dropping after 30 seconds,
Answer() is usually the cause.
Interesting. I can't say that I've ever had that problem.
b.
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On Sat, 2008-04-26 at 18:41 -0400, Andreas van dem Helge wrote:
Does anyone have a script for manual wardialer for asterisk? not sure
if wardialer is the correct term but basically I want to call X
number say 555- through 555-0050 and be able to listen to each
call and when I hang up or
On Thu, 2008-05-22 at 14:05 +0200, nik600 wrote:
No, i'm just wondering because there is creating a greater difference
between my installation and the actual Asterisk.
If it ain't broke, don't fix it. You are already so far behind that any
upgrade is going to be a major task of testing and
On Mon, 2008-05-19 at 11:13 +0100, Adrian Marsh wrote:
Hi Brian,
Thanks for the reply. I tried searching for your posts, but no luck.
I find effective use of the Internet absolutely depends on knowing how
to search for stuff.
On Sat, 2008-05-17 at 18:38 +0100, Adrian Marsh wrote:
All,
Does anyone know of a SIP URI direct to googles 800-GOOG-411 service?
Yeah, I suppose a direct SIP connection would be nice.
An enum lookup shows 3 URIs listed, none of them seem to be google
directly,
No, they are SIP-PSTN
I have a Wildcard FXO: Wildcard X100P (clone) in my Asterisk (1.4.17)
machine and as of late, Caller-ID on it seems to be failing more
frequently than not. Sometimes I get callerid.c:613 callerid_feed:
Caller*ID failed checksum sometimes it fails without even that.
In Zapata.conf I have:
On Thu, 2008-05-08 at 10:51 -0500, Russell Bryant wrote:
Have you taken a look at the ENUMQUERY() and ENUMRESULT() functions that are a
part of Asterisk 1.6?
The ENUMQUERY() function lets you do a single enum query
From a single zone it seems. So that means a for zone in $ZONES type
of
To this end, I have taken a first pass at a Perl AGI script to look up
and return a list of URIs for a given phone number. I will not pretend
that I have read the relevant RFCs but have implemented based on the
knowledge I have gathered about ENUM lookups from various sources.
Given my dialplan
On Thu, 2008-05-08 at 10:51 -0500, Russell Bryant wrote:
Have you taken a look at the ENUMQUERY() and ENUMRESULT() functions that are a
part of Asterisk 1.6?
I have not even entertained thinking of 1.6 yet. :-/
The ENUMQUERY() function lets you do a single enum query for a number. Then,
On Wed, 2008-05-07 at 14:26 +0200, Johansson Olle E wrote:
Quoting RFC 3824:
Only one SIP URI, ideally, appears in an ENUM record set for a
telephone number. While it may initially seem attractive to
provide multiple SIP URIs that reach the same user within ENUM,
if
On Wed, 2008-05-07 at 08:21 -0400, Matt Watson wrote:
There is a enumlookup.agi that is included with FreePBX and thus trixbox, PBX
in a flash, etc. etc.
Yeah, I had gotten that impression somewhere too.
If you have trouble finding it let me know and I can send you it.
If you would be so
On Wed, 2008-05-07 at 22:54 +0300, Tzafrir Cohen wrote:
Slightly off-topic:
Yeah.
On Wed, May 07, 2008 at 10:29:47AM -0400, Brian J. Murrell wrote:
I guess a code audit will tell. :-) Although I got an impression that
it was written in PHP. I'm not much of a fan of PHP. Don't really
On Wed, 2008-05-07 at 13:40 -0700, John Todd wrote:
1) The ENUMLOOKUP function is currently being fixed for TRUNK.
Ahhh. Sweet. I wonder how difficult a backport will be.
Take a look at http://bugs.digium.com/view.php?id=8089 for the
current status. Testing would be appreciated.
Will
On Mon, 2008-05-05 at 16:36 -1000, Julian Yap wrote:
That was a bug in the release.
From the 1.4.20-rc1 Changelog:
2008-04-30 16:30 + [r114891] Russell Bryant [EMAIL PROTECTED]
So basically, r114891 was a fix to AST-2008-006? So if you applied the
patch for AST-2008-006 you now really
On Tue, 2008-05-06 at 13:23 +0100, Julian Lyndon-Smith wrote:
Yes.
Hrm. For those of us that are following along the AST-* train, patching
as per the AST-* release notices, as a matter of process, wouldn't it
have been good to republish AST-2008-006 and include this fix along with
the original
On Tue, 2008-05-06 at 08:42 -0500, Tilghman Lesher wrote:
It's not actually a fix to the security fix.
No, indeed.
The security fix simply
highlighted an issue which was already present in Asterisk.
That may be true, but the security fix now depends on that new fix, so
it's tangentially
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