In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi all,
the default message for email notification looks like:
Is there something wrong with my config?
thx in advance
This should work. Have you reloaded Asterisk?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Hi users;
i am new in the mailing list and asterisk user . i
have to implement METHOD 3 of the link
(http://www.voip-info.org/wiki/view/Asterisk+at+largeview_comment_id=11963)
i have question that is:
Q:when lets i have getting a NOTIFY message and my
phone changes the tone to a MWI tone now if
Anyone can help me to solve the problem about playing the prompt? Is
it related to the package problem? Anyone can give me a clue to find
out the solution? Thx.
I have a simple dial plan to play a voice prompt as follow.
exten = ,1,Answer()
exten =
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
what about this?
show app ringing?
exten = _7XX,1,Ringing
exten = _7XX,2,Goto(local,${EXTEN},1)
It looked promising so I tried it. Unfortunately it didn't help. Calling person
doesn't hear ringing. I don't know why this application
Hello,
Does anybody knows how to make call to leave the queue when all agents in
that queue are busy?
Right now it tries to dial busy members and does not leave queue:
-- Got SIP response 486 Busy Here back from 172.16.2.160
-- SIP/118-082252a8 is busy
-- Called SIP/118
-- Got
Hi,From http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/148565/focus=149455, you can read that :- SIP allows CallerID to be changed at the point when 2 separate calls are bridged to one ...
- May 2006 trunk version of Asterisk did not support this behaviour at that time.Is it still
On Mon, Sep 18, 2006 at 09:39:42PM -0600, Douglas Garstang wrote:
I wonder if the look and feel of this GUI will be completely
configurable. If it's not, then I really don't think that's very
useful. Service providers wouldn't be able to use it to let their
customers manage their own
I have a Sangoma PRI card configured for E1 line (i.e. 30+1 channels per
port) and I'm not quite sure on how the Dial command should performed.
I'm using the standard Dial command as if it were a Zap channel. For example
Dial(Zap/1/1234) ; Dial 1234 using channel 1 on port 1
David Gagnon schrieb:
Are you having this problem with an analog line or PRI ?
David
Sorry, forgot to include that information: It's a PRI.
My Asterisk is: Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q with Zaptel 1.2.6.
Tobias
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Hi Asterisk Users,
I have following problem:
Some external calls from some extensions/nets ( eg. Public phones,
05 ,... ) always reach the -0 extension ( Mainoffice ) although they
dialed some specific extension. In the CDR Table, in the clid and src columns
I see some strange
mediatrix DOES support SIP Register, just enter authentication details and a
registar server
C F wrote:
Keep in mind that the Mediatrix does not support register (AFAIK,
anyhow). You have to create a static entry in sip.conf that has host
set to the IP address of the Mediatrix
On 9/18/06,
I dial using groups. Dial(Zap/g1/1234)
I'm pretty sure this was taken off of the examples on the Sangoma website.
On 9/19/06, Mario [EMAIL PROTECTED] wrote:
I have a Sangoma PRI card configured for E1 line (i.e. 30+1 channels perport) and I'm not quite sure on how the Dial command should
I wonder whether you are experiencing the following bug (since the SIP
INVITE has a multipart SDP body):
http://bugs.digium.com/view.php?id=7124nbn=4
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Dinesh Nair
Sent: 19 September 2006 06:54
To:
Hi all,I'm planing to develop a solution based on Asterisk for about 300 users. My question now is, do I really need to use openSER as the sip proxy and Asterisk for the PBX functions?Can i trust in a solution only with Asterisk to make all this install?
Please help me with your experience on this
On Tue, Sep 19, 2006 at 12:12:03AM +0900, Gary Guthary wrote:
This is going to be an exercise in 'Networking' for sure...
The only catch is that per the phone's network settings:
The phone uses a static IP of something like 192.168.0.220 with a Gateway of
192.168.0.1. - Standard class
Hi,
Asterisk gives me an WARNING if I try to register with my alcatel oxo pbx.
Sep 18 11:27:19 WARNING[11977]: chan_sip.c:5259 check_via: Don't know how
to respond via 'SIP/2.0/udp'
An INVITE to asterisk seems to go through (debug entries in *) but the the
pbx seems to get no SIP responses.
That's ok if I want to dial through a group. But, for my specific
requirements, I need to dial through a specific channel. I even need to
use the ChanIsAvail application to discover which channels are available.
Thus, without using a group, which is the correct way to dial through a PRI?
Hi Ronald,On 9/19/06, Ronald Wiplinger [EMAIL PROTECTED] wrote:
I am thinking if re-invite will interfere accounting.No it won't
Please help me to figure it out:Phone A is registered at asterisk and calls a gateway. If the gatewayallows re-invite than the rtp would go directly from phone A to
On Mon, Sep 18, 2006 at 05:07:31PM -0700, George Pajari wrote:
Any thoughts on this one?
IBM xSeries 330 processor running Debian 3.1 (2.6.8 kernel) with a
TE406P board.
Working fine (more or less) connected to a couple of PRIs.
Rebuild kernel with support for second CPU and inbound
I think that (one of the) offending line(s) is in chan_sip.c:
if (strcasecmp(via, SIP/2.0/UDP)) {
ast_log(LOG_WARNING, Don't know how to respond
via '%s'\n, via);
return -1;
}
This is looking for an upper-case 'UDP'
Oh, when I said offending line I didn't mean to imply that Asterisk is
wrong - I think that the OXO PBX should be using upper-case. Sorry.
-Original Message-
From: Steve Langstaff
Sent: 19 September 2006 10:18
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
hi,
I did it like this:
I wrote a PHP AGI script, that I call from the dial plan.
In the AGI I check
fwrite(STDOUT,CHANNEL STATUS $currchan \n);
fflush(STDOUT);
where currchan is ZAP/1 for instance. It returns whether the channel is used. I then pass this
back as a variable back to
On Tue, 2006-09-19 at 02:18 -0700, Steve Langstaff wrote:
I think that (one of the) offending line(s) is in chan_sip.c:
if (strcasecmp(via, SIP/2.0/UDP)) {
ast_log(LOG_WARNING, Don't know how to respond
via '%s'\n, via);
return -1;
ysuf,
that's exactly what I'm doing (in Python instead of PHP, but that
doesn't matter). However, my question is: should I ask if ZAP/1 is
available or if ZAP/1-1 is available? For example:
ChanIsAvail(Zap/1Zap/2Zap/3)
or
ChanIsAvail(Zap/1-1Zap/1-2Zap/1-3)
And, once discovered
Natambu Obleton wrote:
Ok. First question is how to make it say my number back.
Like if you call extension 1000 from extension 1001, I want it to say
“Number is 1,0,0,1” like an ANI number? Help.
Also I want to setup a meetme conference so that it asks “Enter
conference
Marco Mouta wrote:
Hi all,
I'm planing to develop a solution based on Asterisk for about 300 users.
My question now is, do I really need to use openSER as the sip proxy and
Asterisk for the PBX functions?
Can i trust in a solution only with Asterisk to make all this install?
Please
Mario,
try ChanIsAvail(Zap/1-1)
but when you dial, its Zap/1/${EXTEN}
HTH
Mario wrote:
ysuf,
that's exactly what I'm doing (in Python instead of PHP, but that
doesn't matter). However, my question is: should I ask if ZAP/1 is
available or if ZAP/1-1 is available? For example:
I just use Dial(Zap/1/1234)
[]'s
MM
-Original Message-
From: Mario [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Cc:
Sent: Tue, 19 Sep 2006 12:17:07 +0200
Delivered: Tue, 19 Sep 2006 07:06:25
Subject:[asterisk-users] How to Dial a number with Sangoma PRI card?
You should try :
exten = _,n,ChanIsAvail(Zap/XY)
exten = _,n,NoOp(AvailChannel=${AVAILCHAN})
exten = _,n,Set(__DialChannel=${CUT(AVAILCHAN,,1)})
exten = _,n,Dial(${DialChannel}/YOURNUMTODIAL)
Where X stands for the strategy to fill your PRI ( r,R,g,G,.. ) and Y
stands for the
Hi, all
Can I limit calls in one iax2 trunk just like sip peers do?
How?
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Rushowr wrote:
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We're having fun trying to get a Patton (Inalp) SmartNode 2400 to
function as a T.38/PRI gateway with Asterisk handling the pass-through.
Any other SN2400 users out there with forehead-shaped dents in their walls?
--
George Pajari, netVOICE communications604 484 VOIP (484 8647 x102)
Open
I had changed my setup to iaxmodem/hylafax and that is wonderful.
Works like charm. Thanks your help!
I have problem with printing incoming faxes because it's likely skip
the header and the footer on some fax. I receive the fax on email as
well and pdf is perfect so there is some problem (might
Someone already said that they saw it at VON. It was super simple to
change the look and branding but the UI itself was nothing too special.
Thanks,
Steve
Douglas Garstang wrote:
I wonder if the look and feel of this GUI will be completely configurable. If
it's not, then I really don't
Hi Users;
i have to implement MWI scenario like this:
IPphone,ATAopenserAsterisk
my users are registered at openser and voicemail box
is configured at asterisk.
MWI is send by ASTERISK to OPENSER and then OPENSER to
IPPHONE OR ATA.
My query is this;
Q:let say i got a
Ma Zhiyong wrote:
Hi, all
Can I limit calls in one iax2 trunk just like sip peers do? How?
http://www.voip-info.org/wiki/view/Asterisk+cmd+CheckGroup
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor
Artifex Maximus wrote:
I have problem with printing incoming faxes because it's likely skip
the header and the footer on some fax. I receive the fax on email as
I would suggest asking this question on the HylaFAX mailing list.
Doug
--
Ben Franklin quote:
Those who would give up Essential
Oops! Yes, I was confusing strcmp and strcasecmp, sorry - too little
coffee.
However, it's funny that the strcasecmp of SIP/2.0/UDP fails to match
with SIP/2.0/udp. Almost like it's using a strcmp instead of
strcasecmp!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Benjamin Jacob wrote:
Rushowr wrote:
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Hi,I want to setup a VOIP call center that will be also able to send calls to PSTN over TE110P or TE205P.The first question is if i need to go with a clustering architecture (meaning that i am going to need two PCs and two cards) or a single (strong) PC with one card is sufficient?Secondly the
I'm planning to deploy an Asterisk system in our office soon, and am
thinking of using a mixture of Aastra 9133i and Atcom AT-320 phones.
Has anyone got any comments (good or bad) about these phone models?
Thanks,
James
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--Bandwidth and
I've never used the Aastra but the AT-320's seem to work fairly well, my
only bug with them is there lack of weight, they slide across the desk
to readily.
Bails
James Dyer wrote:
I'm planning to deploy an Asterisk system in our office soon, and am
thinking of using a mixture of Aastra
Doesnt anyone know if this is possible?
2006/9/13, Mir [EMAIL PROTECTED]:
Hello
Is there a possibility for sending an event on the managerinterface
(AMI) when a call is put on/off hold?
Or is there any other way to detect when a call is placed on hold?
Michael
Benjamin Jacob wrote:
Rushowr wrote:
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And also it would be nice to have more detailed instructions on how to
upgrade FOP.
This belongs to FOP mailing list... but anyways:
0) backup your previous install, just in case
1) replace op_server.pl
2) replace operator_panel.swf
3) read UPGRADE, you might need to add 3 or 4 new parameters
Format_MP3 appears to play MOH files starting at the
beginning of each file, using the .wav file format, making for some repetitive hold
music unless you alter the file itself to begin somewhere in the middle.
Solution: One stream that all users connect to
giving dynamic hold music
It the question why does asterisk has problems with SIP/2.0/udp or
SIP/2.0/UDP
if (strcasecmp(via, SIP/2.0/UDP)) {
ast_log(LOG_WARNING, Don't know how to respond via '%s'\n, via);
return -1;
}
This code says: I don't know what to do with a SIP/2.0/UDP in a via and
blocks (return
This is really
starting to get to me. I have deleted this field in the phones per the wiki. I
am trying to get the phones to dial on there own. Is there anyway to get the
phone to dial 1-8 after three digits are received and 9 after seven to ten
digits. I am willing to wait for a timeout
Ryan wrote:
Can you explain your design in a little more detail? What kind of hardware
did you use to get over 1k users on a single box and 500 concurrent calls?
Sounds like a very interesting medium-large scale implementation that
others could learn from.
thanks,
Ryan
I'll do the best
On Tue, 2006-09-19 at 13:13 +0100, James Dyer wrote:
I'm planning to deploy an Asterisk system in our office soon, and am
thinking of using a mixture of Aastra 9133i and Atcom AT-320 phones.
Has anyone got any comments (good or bad) about these phone models?
I now only use Aastra phones,
When I press # (once) during a bridged call it mutes/locks the call and I am unable to do anything on either end of the call.Then, if the called party hangs up before the calling party hangs up, the call still shows as connected with (show channels) even though both parties have hung up. Both
Thanks for your help. I'll send my future FOP questions to FOP mailing list. I am still having some troubles, like it doesn't show more than 34 extensions buttons, doesn't matter if button sizes are made smaller. Rest of the extensions don't show anywhere.
I had the same problem.The only solution I've found is to change blindxfer = ## to blindxfer = *#
or something like that.I think there is a bug with that feature. It was working with some previous version but I don't remember which one.MichaelOn 9/19/06,
marvin horst [EMAIL PROTECTED] wrote:
Hi!
I have the following problem: I route calls from one office to the other
office via SIP, but if for any reason this SIP call fails, I want a
backup route via the PSTN.
Thus, I use:
exten = _[1-9].,4,Dial(SIP/${enumresult},90)
exten = _[1-9].,5,GotoIf($[${DIALSTATUS} =
-Ursprüngliche Nachricht-
Von: Klaus Darilion [mailto:[EMAIL PROTECTED]
Gesendet: Dienstag, 19. September 2006 16:03
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] fast SIP failover (outgoing sIP requests) with
1.2
Hi!
I have the following problem: I route calls
Guido Hecken wrote:
-Ursprüngliche Nachricht-
Von: Klaus Darilion [mailto:[EMAIL PROTECTED]
Gesendet: Dienstag, 19. September 2006 16:03
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] fast SIP failover (outgoing sIP requests) with
1.2
Hi!
I have the following problem: I
Hall, Eric M. wrote:
I got the config working. Not sure if someone has pre-recorded sounds
for this app or not. Looked all over for them and I'm unable to locate
them.If anyone has sound file they would like to share that would help
me greatly.
Thanks
*Sent:* Friday, September
Per 2.0.1 release notes
13315: Increased the maximum number of buddies to 8 for all platforms
except
SoundPoint IP 600 and 601 which can watch 48 buddies
On Sep 18, 2006, at 10:35 PM, Douglas Garstang wrote:
48 was the limit on the number of speed dial entries that you could
have in the
the digitmap only tells the phone when to send the digits it has
collected. They have no digit substitution feature. This would be
done within your * dialplan
On Sep 19, 2006, at 7:57 AM, Jordan Novak wrote:
This is really starting to get to me. I have deleted this field in
the phones
Anyone encountered this on yet?
WARNING[23251]: chan_sip.c:2570 sip_write: Asked to transmit
frame type 64, while native formats is 4 (read/write = 4/4)
Started after an upgrade from CVS 8/2005 to current 1.2.12.1
If I had a reference for what frame types 4 and 64 are I might
be
Rushowr wrote:
Hall, Eric M. wrote:
I got the config working. Not sure if someone has pre-recorded sounds
for this app or not. Looked all over for them and I'm unable to locate
them.If anyone has sound file they would like to share that would help
me greatly.
snip--PLEASE learn to trim
We tested a couple 9133i, dont remember the specifics right now but
we stopped as there was some inconsistency in provisioning. I was
very optimistic as I like the look and feel. We did deploy a couple
480iCT which worked very well - when they worked. But they keep
locking up and freezzing
On 16:44, Tue 19 Sep 06, Klaus Darilion wrote:
Hi!
I've tried it but apparently chanisavail does not work with non-local
SIP peers.
In sip.conf try to add qualify=yes to the remote office part
This will make the sip peer dissapear when it's more then 2
seconds away (probably down). This
Damon Estep wrote:
Anyone encountered this on yet?
WARNING[23251]: chan_sip.c:2570 sip_write: Asked to transmit frame type
64, while native formats is 4 (read/write = 4/4)
Started after an upgrade from CVS 8/2005 to current 1.2.12.1
If I had a reference for what frame types 4 and
Could a carrier use it to provide service to it's customers, such that each
customer only had access to their own content?
-Original Message-
From: Steve Totaro [mailto:[EMAIL PROTECTED]
Sent: Tue 9/19/2006 5:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Try taking to 90 second timeout off
Change
exten = _[1-9].,4,Dial(SIP/${enumresult},90)
to
exten = _[1-9].,4,Dial(SIP/${enumresult})
a btter method is to set up each office as a unique peer with qualify = yes and
then add the peer name to the dial string, like dial(SIP/[EMAIL PROTECTED])
if
I had read a post somewhere that there is an XML parameter for
the Polycom config files for default handset volume, but I can not locate it
again.
Anyone know what it is?
I want to set the default handset volume higher on some
phones, despite the ADA hearing aid warning in the admin
SM Sorry, should have been a little more specific. I've had Asterisk running realtime SIP users/peers andSM realtime sql calls from the dialplan (all with MySQL), and have had around 2.5k registered users and a
SM peak (that I recall) of around 500 concurrent calls.Wow that sounds pretty neat.
Michiel van Baak wrote:
On 16:44, Tue 19 Sep 06, Klaus Darilion wrote:
Hi!
I've tried it but apparently chanisavail does not work with non-local
SIP peers.
In sip.conf try to add qualify=yes to the remote office part
This will make the sip peer dissapear when it's more then 2
seconds away
Hello - I'll be heading out to the Boston area next week to start up a
branch office for my company. I'll be implementing an Asterisk box as
part of their network infrastructure...so...does anyone have any
recommendations on a good reliable SIP or IAX provider? I'd need DIDs
for incoming calls
[EMAIL PROTECTED] wrote:
Can you explain your design in a little more detail? What kind of hardware
did you use to get over 1k users on a single box and 500 concurrent calls?
Sounds like a very interesting medium-large scale implementation that
others could learn from.
thanks,
Ryan
Never mind my previous question. I see that Ryan already asked that and you responded. It does sound like a pretty neat project. Would love to hear more once you finish the project that you are working on currently.
- AK
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Hi all
I've got the following message from the telco regarding call forward
number presentation. Can someone please help me decipher this? I
don't understand shit about this :P
roy
Roy ,
Below is an extract taken from a working scenario of the CFU (A B -
B C) functionality problem :
I've also found similar problems with blindxfer -- such as when someone is attempting to interact with an IVR using a '#' option. By default # seems to transfer a call, but if you have blindxfer enabled with '#1' or ##, then Asterisk hears the first # and waits for a following DTMF.
At the moment
I don't think you can set a default volume, but you can configure the handset
(and headset) volume to persist between calls. Look for 'persist' in sip.cfg or
phone1.cfg.
Doug.
-Original Message-
From: Damon Estep [mailto:[EMAIL PROTECTED]
Sent: Tue 9/19/2006 9:23 AM
To:
I am talking about the GUI that was announced as part of the new Asterisk
Appliance.
Sounds like it is going to be a full featured GUI like FreePBX.
-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Monday, September 18, 2006 8:11 PM
To: Asterisk Users Mailing List
I think the following should resolve your problem, if I'm understanding it correctly:In the sip.cfg file, look at setting the following:volume voice.volume.persist.handset="1" voice.volume.persist.headset="1" voice.volume.persist.handsfree="1"/1= remember last setting, 0=return to defaultThis will
Anthony Cennami wrote:
I've also found similar problems with blindxfer -- such as when someone is
attempting to interact with an IVR using a '#' option. By default # seems
to transfer a call, but if you have blindxfer enabled with '#1' or ##, then
Asterisk hears the first # and waits for a
On Tue, Sep 19, 2006 at 02:53:17PM +0200, Christian Gatti wrote:
It the question why does asterisk has problems with SIP/2.0/udp or
SIP/2.0/UDP
if (strcasecmp(via, SIP/2.0/UDP)) {
ast_log(LOG_WARNING, Don't know how to respond via '%s'\n, via);
return -1;
}
This code says: I
I've heard a MOS of 4.4 is ulaw/alaw, presumably like a local exchange
call through the pstn.
testyourvoip.com tells me that the highest score available with G.729 is
4.2, which is pretty darn close to 4.4.
I don't know why I think this (or why I've heard it (or if it's right))
but I think
Hi
all,
I'm using * 1.0.9
which use mpg123 for music on hold. But sometimes starts eating up a lot of
CPU.
I sthere any
alternative method to use moh without use mpg123?
I tryied this http://astrecipes.net/?n=152but i
doesn't wotks for me.
Anyone can help me
pls ?
Thanks in
advance.
Which we use on all calls so that if someone wants to transfer said in/out call they can (such as a secretary placing a call for an executive, or a regular user transfering a call to a verification service/person.)
On 9/19/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Anthony Cennami wrote:
SM Sorry, should have been a little more specific. I've had
Asterisk running realtime SIP users/peers and
SM realtime sql calls from the dialplan (all with MySQL), and
have had around 2.5k registered users and a
SM peak (that I recall) of around 500 concurrent calls.
Wow that sounds
Hi Folks,
We're trying to roll Asterisk out to production and are having a few
complications.
Most specifically we have G711 for our inbound origination, but would
prefer G729 for outbound termination, so far so good - it appears that
dtmfmode=auto works in both cases.
The area I'm having
Anyone know how to setup the SIP lines on a Citel box so it can register
with Asterisk. I keep getting Unauthorized and I have tried every
different combination of settings that I can think of. I am not sure
what fields are required or what information goes where in the Citel
interface.
Hello,
I wish asterisk to forward the none local uri to an
outbound proxy instead of send back 404 .
I add this in extension.conf and outbound is a peer
with an outbounproxy set to SER :
[sip]
exten = _.,1,NoOp(Incoming Call from house extension
${CALLERID} for [EMAIL PROTECTED])
exten =
On Tue, Sep 19, 2006 at 07:01:41PM +0200, Roy Sigurd Karlsbakk wrote:
The tests we've done shows that asterisk doing RTP bridging SIP/SIP
calls can handle up to approxmately 4-500 calls for a single Xeon 3.0
before locking up, spending approx 60-70% system/kernel time, _not_
usertime. We
On 9/19/06, Erik Anderson [EMAIL PROTECTED] wrote:
Hello - I'll be heading out to the Boston area next week to start up a
branch office for my company. I'll be implementing an Asterisk box as
part of their network infrastructure...so...does anyone have any
recommendations on a good reliable SIP
When Asterisk (1.2.12.1) receives a SIP register message for a realtimepeer, the CLI reports "Disconnected from Asterisk server". Asterisk hasdisappeared:asterisk -rUnable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctlexist?)A look at the full log doesn't reveal much:Sep 17
Use type=user for inbound and type=peer for outbound. Have different
codec settings for each of them.
Mr. Jones wrote:
Hi Folks,
We're trying to roll Asterisk out to production and are having a few
complications.
Most specifically we have G711 for our inbound origination, but would
prefer
Mojo with Horan Company, LLC wrote:
I've heard a MOS of 4.4 is ulaw/alaw, presumably like a local exchange
call through the pstn.
testyourvoip.com tells me that the highest score available with G.729
is 4.2, which is pretty darn close to 4.4.
Alaw and ulaw are about 4.4. ulaw on a robbed
Gang,With the latest code I am having audio issue with the chan_gtalk and wondering if anyone else had the same problem- I was able to set up the call between Asterisk and my gTalk account, but there was no audio
- Looking closer, I am seeing these messages for an incoming call: --
I should have posted the logs when the call is acceptedhere it is:-- SIP/5001-081ef020 answered Gtalk/guan.alex-e086[Sep 19 12:13:47] DEBUG[31226]: rtp.c:1402 ast_rtp_early_bridge: Channel 'Gtalk/guan.alex-e086'
has no RTP, not doing anything[Sep 19 12:13:47] NOTICE[31226]: chan_gtalk.c:1331
Your phones do not have a native transfer feature?
Anthony Cennami wrote:
Which we use on all calls so that if someone wants to transfer said in/out
call they can (such as a secretary placing a call for an executive, or a
regular user transfering a call to a verification service/person.)
On
i have trixbox running, the latest version and when i make an outgoing call from this phone it doesn't pick up the user's name, but instead display the number. is this a grandstream problem? or asterisk? i did specify the user name from 'extension" within trixbox. in matter of face, if i call
test it with someextension and using a sipphone like xlite or something else and you will be able to understand if it is a grandstream issue!hope it helpsOn 9/19/06,
Christopher Corn [EMAIL PROTECTED] wrote:
i have trixbox running, the latest version and when i make an outgoing call from this
marco, can you explain what it is your recommending? i dial now from SIP phone to SIP phone. one being a grandstream gxp grandstream 100. but the gxp, when dialing, doesn't see the name of the person at the grandstream 100. i believe this should be picked up from the asterisk server, but im not
I was looking through my log files and keep seeing:
PRI got event: 8 on Primary D-channel of span 3
on different spans. sometimes it is event 6.
Does anyone know what causes this?
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On 9/19/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Your phones do not have a native transfer feature?Our phones are a mix of Zap and SIP channels
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asterisk-users mailing list
To
I don't see a -- Saved useragent line for the
SPA 3102 device am trying to connect to Asterisk.
I have similar configuration for the SPA 3102 as I
have for another hard phone in the sip.conf file but
the device (SPA 3102) does not even attempt to
register.
I have configured the device to
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