On Thursday 02 July 2009 10:02:03 am David Backeberg wrote:
On Tue, Jun 30, 2009 at 10:47 AM, Ade
Vickersaster...@solutionengineers.com wrote:
I run a phone in a remote office using the IAX2 protocol. It mostly works
fine; except that every 5 mins it loses connection with Asterisk, before
Hello!
Thank you for that piece of information. Which RFC does it state that
the audio name is G729?
Thanks,
Elliot
On Thu, Jul 2, 2009 at 12:16 AM, Kevin P. Flemingkpflem...@digium.com wrote:
Elliot Murdock wrote:
Hello!
I have a sip device that is sending in the SDP:
rtpmap:98 g729a
folk,
I see from the archives that the issue of nortel handsets not
sending dtmf tones to asterisk has been discussed a couple of times,
but there is something I quite havent seen answered yet.
Is this dtmf issue a problem with the nortel handsets or the PBX
itself? If the handset were changed
Elliot Murdock schrieb:
Thank you for that piece of information. Which RFC does it state that
the audio name is G729?
http://tools.ietf.org/html/rfc3555#section-4.1.9
On Thu, Jul 2, 2009 at 12:16 AM, Kevin P. Flemingkpflem...@digium.com wrote:
Elliot Murdock wrote:
I have a sip device that
Hello!
I noticed that the SIP packet contains this line:
m=audio 6 RTP/AVP 18 98 96 97 101 13
However, there is no rtpmap that describes 18. Media format 18
Apparently refers to G729, but there is no rtpmap in the SDP for it.
Since G729 is a registered and known format is there any way for
Hello!
Oops, a typo:
Hello!
I noticed that the SIP packet contains this line:
m=audio 6 RTP/AVP 18 98 96 97 101 13
However, there is no rtpmap that describes 18. Media format 18
Apparently refers to G729, but there is no rtpmap in the SDP for it.
Since G729 is a registered and known
Hello,
I use asterisk as a PRI gateway between the PRI line from the telco
(span 1) and the customers PBX (span 2) . I did a pri debug because I
had some problems with incoming routings.
From the telco I get:
1 Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive
Dchan: 0
1
Hello Phillip!
Thank you.
However, do you what Asterisk does if there is no rtpmap that
describes the format profile in the media description?
The SIP packets states:
audio 25184 RTP/AVP 18 98 96 97 101 13
However, there is no rtpmap for format 18. Asterisk does not seem to
be associating 18
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Ernest Byaruhanga
Sent: Thursday, July 02, 2009 4:10 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Nortel pbx dtmf issues
folk,
or maybe i misread :)
2009/7/1 Mike Dent mcd...@gmail.com
2009/7/1 Geraint Lee gera...@gmail.com:
agreed.
extended o2 coverage would be very useful, especially for Wales!
I like the idea, i don't like the idea of paying, if they want mobile
traffic it should be possible to buy your
Dear Sir,
I have a Patton device registered on an OpenSIPs server...The openSips
Server send the call to asterisk serverWhen using some end user device
like patton the asterisk server does not send back an Authentication
required packet to INVITE packet received from OS as follow:
--- SIP
Actually it was my Firewall (Endian). By rebooting my firewall, all
problems were solved and till this moment every communication succeeds.
I do expect them back...
I don't want to hijack this Asterisk-mailinglist, but I think that
firewall-issues also are related to Asterisk-support.
So my
Just by blocking the packets coming from 3stars.
Your asterisk, by receiving no responses think thet the host is not
reachable.
They are mayny good routers/firewall permitting to avoid these kind of
problems.
Olivier
jonas kellens a écrit :
Actually it was my Firewall (Endian). By
Philipp Kempgen wrote:
http://tools.ietf.org/html/rfc3555#section-3
---cut---
Note that the payload format (encoding) names defined in the RTP
Profile are commonly shown in upper case. MIME subtypes are commonly
shown in lower case. These names are case-insensitive in both
Elliot Murdock wrote:
Hello!
I noticed that the SIP packet contains this line:
m=audio 6 RTP/AVP 18 98 96 97 101 13
However, there is no rtpmap that describes 18. Media format 18
Apparently refers to G729, but there is no rtpmap in the SDP for it.
Since G729 is a registered and
Hello Kevin,
Gotcha...however, the mime is G.729a with an extra period, so it
doesn't get recognized.
However, as I asked before, does Asterisk map any of the RTP/AVP
profiles or does every format need to be defined in the SDP with a
rtpmap attribute?
You see, the incoming SDP supplies format
I've been trying to get an AGI transfer to work for several weeks now. It
isn't error-ing out, but it isn't working either.
I can't use dial in this case due to what I'm trying to accomplish.
Does an AGI Transfer actually work?
-= Info about application 'Transfer' =-
[Synopsis]
Transfer
Hello,
Thank you clarifying that.
However, if that is the case, why is Asterisk sending back PCMU
packets (instead of G729), which the device is not enabled for and
subsequently, fails the call?
Could the mapping be disabled or not properly mapping to the G729
driver in a certain versions of
Elliot Murdock wrote:
However, if that is the case, why is Asterisk sending back PCMU
packets (instead of G729), which the device is not enabled for and
subsequently, fails the call?
Could the mapping be disabled or not properly mapping to the G729
driver in a certain versions of Asterisk?
How about putting nat=yes for everything even if it isn't NAT, use
qualify, and get rid of your firewall rules if you are registering.
On Thu, Jul 2, 2009 at 8:18 AM, hh174oliv...@hh174.be wrote:
Just by blocking the packets coming from 3stars.
Your asterisk, by receiving no responses think
Hello!
Which RFC specifies the corresponding number of the formats?
Where in the Asterisk source code does it state the SDP formats?
Does Asterisk follow the formats of IANA?
(http://www.iana.org/assignments/rtp-parameters)
Thank you,
Elliot
On Thu, Jul 2, 2009 at 3:44 PM, Elliot
On Thu, 2 Jul 2009, Elliot Murdock wrote:
Hello!
Which RFC specifies the corresponding number of the formats?
Where in the Asterisk source code does it state the SDP formats?
Does Asterisk follow the formats of IANA?
(http://www.iana.org/assignments/rtp-parameters)
Thank you,
Elliot
Just do your make menuselect and make install commands again, making sure
you select sounds in the selection process.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Wednesday, July 01, 2009
Hello Jeff,
Yes, I use G729 all the time.
Here is the SDP extrace from Wireshark. I'll get more data as it
becomes available:
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): MG4000|2.0 49743 83164 IN
IP4
On Wed, Jul 1, 2009 at 7:03 PM, bilal ghayyadbilmar...@yahoo.com wrote:
Hi All;
I download asterisk, compiled it and install it, but not finding the sounds
file (/var/lib/asterisk/sounds), what could be the reason and how I can have
it without repeating every thing?
My asterisk version
On Thu, 2 Jul 2009, Elliot Murdock wrote:
Hello Jeff,
Yes, I use G729 all the time.
Here is the SDP extrace from Wireshark. I'll get more data as it
becomes available:
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id
On Thu, 2 Jul 2009, Jeff LaCoursiere wrote:
On Thu, 2 Jul 2009, Elliot Murdock wrote:
Hello Jeff,
Yes, I use G729 all the time.
Here is the SDP extrace from Wireshark. I'll get more data as it
becomes available:
Session Description Protocol
Session Description Protocol
Elliot Murdock wrote:
Which RFC specifies the corresponding number of the formats?
Have you heard of Google? It's amazing in that it can find stuff for you
like that faster than you can post to a mailing list and wait for a
response :-)
Where in the Asterisk source code does it state the SDP
Jeff LaCoursiere wrote:
Also - please post the output of the CLI command:
core show translation
That won't affect the SDP negotiations; if sip.conf allows g729 for that
endpoint, Asterisk will negotiate it, regardless of whether there is a
transcoding path available or not. Asterisk cannot
On Wednesday 01 July 2009 05:35:08 Tom O'Connor wrote:
On Wed, Jul 1, 2009 at 11:08 AM, Steve Totaro wrote:
That is one option. The new line Digium cards are on par with Sangoma as
far as IRQ issues.
I don't really know what you mean about the new line Digium cards.. which
models are in
Hi All,
A couple of customers called complaining that folks were dialing into
their PBX trying to use the Directory to locate users, from a
Blackberry, and getting frustrated due to the incompatibility of
dialing alpha characters on the the qwerty keyboard and not getting
through.
The issue of
Hello!
Here is the 200 OK response from my server:
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 10646 10646 IN IP4 212.80.21.238
Owner Username: root
On Thu, Jul 02, 2009 at 09:53:18AM -0500, JR Richardson wrote:
Hi All,
A couple of customers called complaining that folks were dialing into
their PBX trying to use the Directory to locate users, from a
Blackberry, and getting frustrated due to the incompatibility of
dialing alpha
Elliot Murdock wrote:
Any help resolving this is would be great!
Even though we've asked, you have not given us what we asked for. The
most important thing to debug this problem is a complete console log,
with 'core set verbose 10', 'core set debug 10' and 'sip set debug on',
and ensure the the
On Wed, Jul 1, 2009 at 6:58 PM, Steve Totaro
stot...@asteriskhelpdesk.comwrote:
On Wed, Jul 1, 2009 at 6:35 AM, Tom O'Connor t...@twinhelix.org wrote:
On Wed, Jul 1, 2009 at 11:08 AM, Steve Totaro
stot...@first-notification.com wrote:
On Wed, Jul 1, 2009 at 5:58 AM, Tom O'Connor
-- Forwarded message --
From: mahboob zaman mahboob.za...@ssl.com.bd
Date: Tue, Jun 30, 2009 at 8:42 AM
Subject: cisco phone 7911
To: asterisk-users@lists.digium.com
Hellow,
I have cisco 7911 and 7906 worked with asterisk server. But i can not set
the time and date for these
No one ever responded to this inquiry but I figured out what the issue
was. I thought I would respond with the solution just in case someone
runs into the same issue in the future.
Firstly, when setting up trunking between servers the username = field
is not optional. :) Also, I had a lot
Hello Everybody!
Here are the full SIP logs!
--- SIP read from 216.48.184.50:5060 ---
INVITE sip:6587972772285...@82.80.231.238:5060;user=phone SIP/2.0
Call-ID: 699864475636237-1246542986-18105
From: sip:7188894...@216.48.184.50:5060;user=phone;tag=24794
To:
On Thu, Jul 2, 2009 at 4:41 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Thu, Jul 02, 2009 at 04:15:13PM +0100, Tom O'Connor wrote:
I have tried all suggestions given. It just happens that none of them
have
been much use. I'm very constrained by time on this project, less than
11
Elliot Murdock wrote:
[Jul 2 16:56:26] VERBOSE[13420] logger.c: --- (12 headers 12 lines) ---
[Jul 2 16:56:26] VERBOSE[13420] logger.c: Sending to 216.48.184.50 :
5060 (no NAT)
[Jul 2 16:56:26] VERBOSE[13420] logger.c: Using INVITE request as
basis request -
Hello!
My Goodness! Thanks for the help!
I did set up a SIP account for that device to allow G729, but it
doesn't seem to be associating the device with that account. So that
is where my confusion came from.
I'll make the changes and see what happens.
Regards,
Elliot
On Thu, Jul 2, 2009 at
I am using 1.4, and have the above device, and it worked really well
with monitoring 18 hints aka devices.
Now, I've moved us to a hotdesking paradigm where the user is the
extension not the device. IOW if I dial 1234, I will get user 1234
(who happens to log on to device ABC today, and DEF
So what I'm gathering is this I have to map each extension to a button,
whether physical on a 560m or 536m or virtual using the soft buttons on
the phone. What I was hoping for was something like the Directory app
http://voip-pbx/aastra/directory.php that came with the phone's firmware
that shows
Why do I see many setups where there is an Asterisk server in
combination with a SIP-server like OpenSER or Kamailio ?
Isn't Asterisk enough as SIP-server ??
It can communicate with many databases through ODBC, with many other
software through an API (AGI), with other servers like OpenFire for
Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The
VM box, itself, is beginning to show its age. Big-time. We're thinking
it
might be time to look for a replacement. I'd love to install Asterisk
with an FXO card or something, but I don't think it supports whatever
On Thu, 2 Jul 2009, jonas kellens wrote:
Why do I see many setups where there is an Asterisk server in
combination with a SIP-server like OpenSER or Kamailio ?
Why is there an extra SIP server implemented at many VoIP-providers ??
OpenSER/Kamailio are supposed to be much better dealing with
On Thu, 2 Jul 2009, J. G. wrote:
I've been trying to get an AGI transfer to work for several weeks now.
It isn't error-ing out, but it isn't working either.
Does an AGI Transfer actually work?
Does executing the transfer application in an AGI using the AGI exec
command work? Yes.
I
hi, i have a new install, 1.6 2, 2 extension, but the call doesnt get
thorugh: here is my sip debug outout: thx for ur help!!
asterisk-users@lists.digium.com
--- (13 headers 16 lines) ---
Sending to AA.BBB.CCC.DD : 28127 (NAT)
Using INVITE request as basis request -
I have had a search for this, but didn't come up with any results, so maybe I
am
using the wrong terms, sorry if this is an FAQ.
For those who want to forward their incoming voice calls to a mobile, it could
be a cheaper option to call a mobile from another mobile on the same network.
This
Are there any good tutorials or overviews on a basic setup using a SIP
router in conjunction with Asterisk? I would love to get a proof of concept
up and working as I'm in the midst of a PBX re-architecture, and having the
load-balancing/high availability features that a SIP frontend would
Rajkiran Reddy wants you to join Yaari!
Is Rajkiran your friend?
a
href=http://yaari.com/?controller=useraction=mailregisterfriend=1sign=YaariLBF849RQF972ZTA396ZMZ718;Yes,
Rajkiran is my friend!/a a
Make sure this is in your xml config:
ntps
ntp
nameip.of.ntp.server/name
ntpModeUnicast/ntpMode
/ntp
/ntps
On Thu, Jul 2, 2009 at 10:27 AM, mahboob zamanmahboob.za...@ssl.com.bd wrote:
-- Forwarded message
Check chan_mobile. Now is mature enough to be used in a server with low CPS.
The USB connectivity will be introduced in the close future (I think) but by
now it can be connected via bluetooth device.
On Thu, Jul 2, 2009 at 3:20 PM, Nick Hill t...@nickhill.co.uk wrote:
I have had a search for
Carlos Ruiz Diaz a écrit :
Check chan_mobile.
[...]
Or use GSM gateway
On Thu, Jul 2, 2009 at 3:20 PM, Nick Hill t...@nickhill.co.uk wrote:
I have had a search for this, but didn't come up with any results, so maybe
I am
using the wrong terms, sorry if this is an FAQ.
For those
On Wed, 2009-07-01 at 07:17 -0700, Jonathan Thurman wrote:
On Tue, Jun 30, 2009 at 11:53 PM, John A. Sullivan III
jsulli...@opensourcedevel.com wrote:
On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan III wrote:
Hello, all. With the assistance of very helpful
Fail.
Rajkiran Reddy wrote:
Rajkiran Reddy wants you to join Yaari!
Is Rajkiran your friend?
Yes, Rajkiran is my friend!
http://yaari.com/?controller=useraction=mailregisterfriend=1sign=YaariLBF849RQF972ZTA396ZMZ718
No, Rajkiran isn't my friend.
Hello list,
I want to know the options that I have as an Asterisk user to make a call
and control its progress.
I have been using the spool directory to originate my calls using this
format:
Channel:
Callerid:
MaxRetries:
RetryTime:
WaitTime:
Context:
Extension:
Priority:
Is there any
On Thu, 2 Jul 2009, Carlos Ruiz Diaz wrote:
I want to know the options that I have as an Asterisk user to make a call
and control its progress.
I have been using the spool directory to originate my calls using this
format:
Is there any other way to originate it and being able to cancel it
Thank you Steve.
AMI is what I used before and now I am looking for something better if
exists.
On Thu, Jul 2, 2009 at 8:20 PM, Steve Edwards asterisk@sedwards.comwrote:
On Thu, 2 Jul 2009, Carlos Ruiz Diaz wrote:
I want to know the options that I have as an Asterisk user to make a call
On Tuesday 30 June 2009 05:01:42 am srinivas Antarvedi wrote:
- To resolve this i tried to remove all keys in all servers and once
again created and
distributed the loaded in each system with keys init command but
stilll i am
getting the same error
can anybody help me out???
Un-top-posting...
On Thu, 2 Jul 2009, Carlos Ruiz Diaz wrote:
I want to know the options that I have as an Asterisk user to make a
call and control its progress. I have been using the spool directory
to originate my calls using this format:
Is there any other way to originate it and
Hello, everyone. No need to read this message. I'm posting for
documentation for other poor, ignorant slobs like me who are struggling
to pull together the many technologies to make converged networks
happen. Hopefully, this will help save someone else the time I spent.
I started the below
On 07/02/2009 10:14 AM, Tzafrir Cohen wrote:
On Thu, Jul 02, 2009 at 09:53:18AM -0500, JR Richardson wrote:
Hi All,
A couple of customers called complaining that folks were dialing into
their PBX trying to use the Directory to locate users, from a
Blackberry, and getting frustrated due to
I finally decided it's been long enough using my ancient HP junker
and I built a Atom 330 based machine to replace it. I've installed
Centos 5, Dahdi and Asterisk 1.6.2.
After a bit of struggles getting the 1.2 version files converted to
1.6 almost everything seems to be working.
All my SIP
It is a problem with Windows mobile phones as well, there is *NO* way
to dial a number e.g. 800-CALL-ATT. On my Nokia S60 phone (E71) I can
dial the number but it is not possible to dial letters when the call
is connected.
This affects everyone. When I call American Express it asks me to
enter my
On Thu, Jul 2, 2009 at 8:51 PM, Steve Edwardsasterisk@sedwards.com wrote:
Un-top-posting...
On Thu, 2 Jul 2009, Carlos Ruiz Diaz wrote:
I want to know the options that I have as an Asterisk user to make a
call and control its progress. I have been using the spool directory
to originate
From: Darrick Hartman
Sent: Thursday, July 02, 2009 8:06 PM
On 07/02/2009 10:14 AM, Tzafrir Cohen wrote:
On Thu, Jul 02, 2009 at 09:53:18AM -0500, JR Richardson wrote:
Hi All,
A couple of customers called complaining that folks were dialing into
their PBX trying to use the Directory to
On Thu, 2 Jul 2009, Steve Totaro wrote:
AMI has historically been flaky under high load.
Not sure about these days, probably better, but once upon a time
pounding on the AMI would crash or just freeze a box, requiring a hard
reboot. The machine would become totally unresponsive except to
On Fri, Jul 3, 2009 at 12:02 AM, Steve Edwardsasterisk@sedwards.com wrote:
On Thu, 2 Jul 2009, Steve Totaro wrote:
AMI has historically been flaky under high load.
Not sure about these days, probably better, but once upon a time
pounding on the AMI would crash or just freeze a box,
Just a thought as we explore the brave new world of converged voice and
emails. Voice mail boxes typically hold a very small number of messages
while email folders contain thousands. Do we need to rethink the
traditionally small limits on voice mail boxes when storing in IMAP or
are the messages
On Thu, 2009-07-02 at 20:59 -0400, John A. Sullivan III wrote:
Hello, everyone. No need to read this message. I'm posting for
documentation for other poor, ignorant slobs like me who are struggling
to pull together the many technologies to make converged networks
happen. Hopefully, this
exten = _X48600XXX,1,MeetMeAdmin(${EXTEN:2},T,${EXTEN:0:1})
exten = _X48600XXX,2,Hangup
exten = _X38600XXX,1,MeetMeAdmin(${EXTEN:2},t,${EXTEN:0:1})
exten = _X38600XXX,2,Hangup
exten = _X28600XXX,1,MeetMeAdmin(${EXTEN:2},m,${EXTEN:0:1})
exten = _X28600XXX,2,Hangup
exten =
On Thu, 2009-07-02 at 17:42 -0400, John A. Sullivan III wrote:
On Wed, 2009-07-01 at 07:17 -0700, Jonathan Thurman wrote:
On Tue, Jun 30, 2009 at 11:53 PM, John A. Sullivan III
jsulli...@opensourcedevel.com wrote:
On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan
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