From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical
Support
Sent: Wednesday, December 03, 2008 11:14 PM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] Windows Mobile 6 SIP client: Remote hostcan't
match request NOTIFY to call
You’ll have to recheck your
Now, I have :
Client 1
-Asterisk1--Asterisk2
Client 2
I need that sip sign go to Asterisk2
But RTP go to Asterisk1 and no more.
Where have I to insert canreinvite ?
Thank you
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Eric
ManxPower
On 3 Dec 2008, at 17:38, BERGANZ François wrote:
Someone have a solution for me ?
De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] De la part de BERGANZ François
Envoyé : mercredi 3 décembre 2008 18:24
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] canreinvite=yes problem
Hi,
December 5th, 1999 was the initial release of Asterisk by Mark
Spencer. We'll be celebrating this by gathering as usual at 12 Noon
Eastern (9AM Pacific, 10 MST, 11 Central, 5PM UK and Western EU) for
the VoIP Users Conference.
You can get all the dial in information at
Hi
I have this in my queue extension and I see this in asterisk when I call to the
queue, but no file is created in the directory any ideas?
exten =
s,1,Set(MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-${UNIQUEID})
-- Executing [EMAIL PROTECTED]:1] Set(SIP/0850001175-b7942770,
You can check the notes/links on the following post, it has a link to an
avaya-asterisk ip trunk setup.
http://www.tek-tips.com/viewthread.cfm?qid=1431673
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David fire
Sent: 03 December 2008
Hi,
A glance at sourceforge.net/projects/openh323 Help Forum made me wonder if
this location is the one to use (I got trouble in the past when google
pointed to an obsolete site) :
some quite old messages remain unanswered.
Cheers
___
-- Bandwidth and
I have thousands if this messages in the logs:
Dec 4 10:53:43 NOTICE[26310]: app_queue.c:1980 wait_for_answer: No one is
answering queue 'COMMERCIAL-WT' (2/0/0)
Dec 4 10:53:43 WARNING[5602]: channel.c:889 channel_find_locked: Warning:
Avoided contention wait for '0xb77482c8', 10 retries! RETURN
As I recall, when openh323.org because obsolete people could download
the PWLib OpenH323 libraries from http://www.voxgratia.org/
Openh323 has moved to become OPAL (supporting SIP, H323 and IAX) and can
be downloaded from http://www.opalvoip.org
H323Plus is also a continuation of OpenH323
Does anyone know of any UK mobile provider who can either provide a
single number for a range of sims, or allow us to change the callerid of
a sim dynamically ? We are looking at between 20-30 sims, perhaps more
next year.
Julian
Dear All,
Thank you for taking the time to read this post - I am *confused!* as to why my
asterisk setup does not work as it should. I have an ISDN 30 connection for
telephony, a Sangoma card, and asterisk installed.
Incoming calls, and outgoing calls work 100%. Making an international call,
On Wed, Dec 3, 2008 at 10:47 PM, JD [EMAIL PROTECTED] wrote:
Grey man: you are right. The direction of a call leg is easy to
determine from the point of view of asterisk. I suspect other folks
however, think of it differently. Some would think of a call coming from
a customer CPE to asterisk
On Thu, Dec 04, 2008 at 11:49:50AM +, Mr Gabriel wrote:
Dear All,
Thank you for taking the time to read this post - I am *confused!* as to why
my asterisk setup does not work as it should. I have an ISDN 30 connection
for telephony, a Sangoma card, and asterisk installed.
Incoming
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, 4 December, 2008 12:01:54 GMT +00:00 GMT Britain, Ireland,
Portugal
Subject: Re: [asterisk-users] BT - ISDN30 - International Calls not working,
everything else is fine :(
-
On 12/4/08, Uros Djokic [EMAIL PROTECTED] wrote:
Hi I have problem with TE121 Digium card. I connected it to modem keymile
Music 200 (provided by telco) but I can see 2 red lights on modem (both
bellow words rx) and my card is red too. I tried to make experiment and made
loopback (pins 1 4
I had front speaker working initially - but have lost that (now back only).
Something isn't quite right - but still workable...
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson
Sent: December 4, 2008 3:10 AM
To: Asterisk Users List
Subject:
Hi All,
Is there any way to provide the user receiving an attended transfer with a tone
or other audible indication that the transfer is completed (i.e. Party A calls
Party B, Party B announces the call while transferring to Party C, Party C
hears tone when Party B completes the transfer so
I still have:
Client 1
-Asterisk1--Asterisk2
Client 2
When client1 do a call, asterisk1 forward to asterisk2, asterisk2 forward to
Asterisk1
At this moment, asterisk1 say : 404Not found
But I have insecure=very
This is the sip debug at that moment:
-
--- (11
On December 2, 2008 07:55:00 pm Erik (Caneris) wrote:
Nuance would say no :)
I'd say maybe. Call up +14164854854, it's a recent project we did for a
That's pretty cool! Is there any SIP or IAX access to this (aside from
dialing a POTS number) ?
-A.
Hi there!
Does anyone deal with Telus in BC ? We have some PRI lines that were used
for dialup, would like to convert them for pbx system, talked with some
technicians @ Telus, but the information given was not clear, kind of: try
this see if it works Does anyone here have the settings
Doesn't look like anyone has any suggestions though, guess it's time to play
until it's fixed then :)
2008/12/2 Thomas Kenyon [EMAIL PROTECTED]
Geraint Lee wrote:
Hello there...
Noticed some strangeness going on with mixmonitor and chanspy, the
called (External SIP) party seem to be
On Wednesday 03 December 2008 19:22:09 Barry L. Kline wrote:
It appears as though * is looking for the temporary greeting on the
local box, which is what I'd expect because of my configuration option.
It also appears that * isn't deleting the file(s) when I ask it to. It
also isn't taking
In article [EMAIL PROTECTED],
Mr Gabriel [EMAIL PROTECTED] wrote:
Thank you for taking the time to read this post - I am *confused!* as to why
my asterisk
setup does not work as it should. I have an ISDN 30 connection for telephony,
a Sangoma
card, and asterisk installed.
Incoming
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tilghman Lesher wrote:
You might want to try 1.6.0.3-rc1, released yesterday. In the fixes were
included something similar to this. In case 1.6.0.3-rc1 does not fix this
issue, please open a report on http://bugs.digium.com.
I don't have the
Hello,
I'm receiving some traffic from a Softwitch to Asterisk
When I'm hiding the CallerID in the softwitch, everything is all right.
When I allow to send the callerid from softwitch to Asterisk (actually,
I would like to have it) Asterisk rejects the call with a 407 Proxy
Authentication SIP
There is a loopback somewhere on the line. Contact your telco and say
I see a loopback on the line. Please remove it.
Uros Djokic wrote:
Hi I have problem with TE121 Digium card. I connected it to modem keymile
Music 200 (provided by telco) but I can see 2 red lights on modem (both
bellow
Reinvites will happen by default. Post your sip.conf [general] and the
peers in sip.conf masking only the passwords. Also paste the part of
extensions.conf that you use to Dial.
BERGANZ François wrote:
Now, I have :
Client 1
-Asterisk1--Asterisk2
Client 2
I need that sip sign
In article [EMAIL PROTECTED],
Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote:
There is a loopback somewhere on the line. Contact your telco and say
I see a loopback on the line. Please remove it.
I don't think this is correct. The OP below said that he put the loopback
on himself, as a
I would try setting zaptel to pri_net before calling the telco.
If it works, you just saved yourself (possibly) hours of being bounced
around from person to person and sitting on hold, not to mention being
on hold or transfered and getting dropped and having to start all over
again.
Path of
Some telco switches just behave this way, either by design or
misconfiguration. It is much easier to reconfigure your switch ;-)
than the telco's.
Not sure what switch Quest had me on, it may have been a DMS100 but I
don't recall. Anyways, pri_net worked and has been working for over
two years
On Thu, 4 Dec 2008, Steve Totaro wrote:
I would try setting zaptel to pri_net before calling the telco.
If it works, you just saved yourself (possibly) hours of being bounced
around from person to person and sitting on hold, not to mention being
on hold or transfered and getting dropped
On 12/4/08, Steve Edwards [EMAIL PROTECTED] wrote:
On Thu, 4 Dec 2008, Steve Totaro wrote:
I would try setting zaptel to pri_net before calling the telco.
If it works, you just saved yourself (possibly) hours of being bounced
around from person to person and sitting on hold, not to
Next time I'll be sure to finish my morning coffee before posting. 8-)
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote:
There is a loopback somewhere on the line. Contact your telco and say
I see a loopback on the line. Please remove
Hi,
How do I disable asterisk to use database and storage voicemail in
directory?
Im getting the below error
[Dec 3 19:08:53] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to
obtain database object for 'asterisk'!
[Dec 3 19:08:55] WARNING[5129]: app_voicemail.c:2353
John:
However, that doesn't mean that it shouldn't be implemented. This is
an area in which I think there is a disproportionate amount of non-
discussion, since many people who would use or be interested in MRCP
simply don't participate in the Asterisk project because it doesn't
meet their
I think its the entries in the extconfig.conf file
Mos
2008/12/4 Geraldo Coelho [EMAIL PROTECTED]
Hi,
How do I disable asterisk to use database and storage voicemail in
directory?
Im getting the below error
[Dec 3 19:08:53] WARNING[4934]: app_voicemail.c:3430 inboxcount:
Thanks. Unfortunately no SIP/IAX access at this time, only by dialing one of
the TNs. However, I'll bring it up with the client and see if they'd want us to
configure that.
Somewhat off-topic, but I'll mention briefly that it's a multi-city service and
you can get more info at
On December 4, 2008 02:14:52 pm Erik (Caneris) wrote:
Thanks. Unfortunately no SIP/IAX access at this time, only by dialing one
of the TNs. However, I'll bring it up with the client and see if they'd
want us to configure that.
Definitely would be cool, you don't lose any ad revenue and I don't
Thanks Kevin,
After looking at the skeleton application, I inserted the line
*AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY,My Prepaid Application);*
To the end of the application. I also changed all references to
pbx_exec(chan,app,data) to include three parameters instead of four (I guess
the APIs
Not work!!!
I am receiving message in my email, but I can't heard from phone because
the asterisk isn't salve message in directory.
thanks
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Mosiuoa Tsietsi
Enviada em: quinta-feira, 4 de dezembro de 2008 17:09
Para:
Someone could make it work???
I tried everything and there's no way I can make it work!
Someone can help me?
Thanks!
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or
On Thursday 04 December 2008 12:46:53 Geraldo Coelho wrote:
How do I disable asterisk to use database and storage voicemail in
directory?
Im getting the below error
[Dec 3 19:08:53] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to
obtain database object for 'asterisk'!
Recompile
So your sendmail is working fine. Asterisk should plug the voicemail in the
/var/spool/asterisk/voicemail/${yourcontext} directory. I would check the
permissions to make sure asterisk has rights to write here.
Mos
2008/12/4 Geraldo Coelho [EMAIL PROTECTED]
Not work!!!
I am receiving
Hi,
I'm using app_pppd with a Digium-PRI-card for PPP connections.
I had some strange problems with some IP packets passing
and some not, e.g. ftp login went well, but as soon as
I tried to up- or download a file, noting was transferred.
I finally guessed, it must have to do something with the
Thanks Mos and Tilghman,
I recompiled it and now is working fine
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Tilghman Lesher
Enviada em: quinta-feira, 4 de dezembro de 2008 18:52
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto:
Welcome to the world of FreePBX. It would save me quite a bit of time
if you could list what ports (port number and signaling) you have on the
card and what context you want each port to go into. When I manually
merge the two files (after stripping out 37 billion comment lines) I see
that
ICMP is used to determine maximim packet size. If you or the other end
are blocking all ICMP then MTU Path Discovery will not work. It's a
classic newbie network admin mistake. Symptoms of this problem would be
exactly like you describe.
Typically I see this on PPPoE connections.
More
Eric \ManxPower\ Wieling schrieb:
ICMP is used to determine maximim packet size. If you or the other end
are blocking all ICMP then MTU Path Discovery will not work. It's a
Hi,
the problem is, the other side (ISDN-router) does not negotiate
the MTU while setting up PPP. I can see this in
Was messing with a polycom 501 and changed the IP from dhcp to static. Working
with a user remotely. Now, the user says the phone does not show anything on
the LCD and does not respond to any buttons.
When rebooting, there is text shown as it proceeds. ??
Is there a way to reset this to a
Is there another web front end for meetme apart from Web-MeetMe? Since
it keeps crashing I need a stable solution for a customer. Any
recommendations? Even a commercial app would be acceptable as long as
it is stable and uses Asterisk.
--
Telecomunicaciones Abiertas de México S.A. de
On 21/11/2008 6:47 a.m., Lincoln King-Cliby wrote:
Hi All,
I have a ticket open with Digium, but based on their previous lack of support
for the Asterisk Appliance, I'm not really holding my breath - and, honestly,
I'm not 100% convinced it's a Digium issue in the first place (but I don't
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, December 04, 2008 7:24 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] polycom no menu
Was messing with a polycom 501 and changed the IP from
We often find ourselves reading through all sorts of contests on the
Internet that never seem to echo our own personal skill set or interests.
Perhaps you've even fantasized about a type of contest with the types of
prizes and goodies that YOU'D actually enjoy. Maybe you've wished there were
Odd problem, where some remote phones, at users homes, dial and connect fine,
no matter what the destination is.
Bad phones, SIP to SIP, between remote and office, or remote to remote, work
and have good audio, but no audio, at all, to PSTN or Cell phones.Phone can
be moved to office and
Ralf Träskman wrote:
Hi
I have this in my queue extension and I see this in asterisk when I call
to the queue, but no file is created in the directory any ideas?
exten =
s,1,Set(MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-${UNIQUEID})
-- Executing [EMAIL
randulo wrote:
Hi,
December 5th, 1999 was the initial release of Asterisk by Mark
Spencer. We'll be celebrating this by gathering as usual at 12 Noon
Eastern (9AM Pacific, 10 MST, 11 Central, 5PM UK and Western EU) for
the VoIP Users Conference.
You can get all the dial in information at
2008/12/4 Vlasis Hatzistavrou (KTI) [EMAIL PROTECTED]
As I recall, when openh323.org because obsolete people could download
the PWLib OpenH323 libraries from http://www.voxgratia.org/
Openh323 has moved to become OPAL (supporting SIP, H323 and IAX) and can
be downloaded from
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