hi:
yes, it is true, the channel is locked by some tasks. you can please restart
asterisk and zatpel.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
Date: Tue, 24 May 2011 00:08:48 +0300
From:
I don't think so. I think it's a predefined pairs of TON and NPI, that
could not be set separately.
On 24.05.2011 04:35, Rafael dos Santos Saraiva wrote:
did not work!! Bug in Asterisk?? :(
Rafael
2011/5/20 Захаров Антон ins...@mail.ru mailto:ins...@mail.ru
Yeap, I couldn't set Private
http://code.google.com/p/zaphfc/
2011/5/23 Patrick Lists asterisk-l...@puzzled.xs4all.nl:
Hi,
I would appreciate some advice on the following: how does one use a single
BRI port HFC chipset based ISDN cards with the latest DAHDI, libpri and
Asterisk 1.8?
For Asterisk 1.4 I would first
23 maj 2011 kl. 23.36 skrev Paul Belanger:
On 11-05-23 05:30 PM, Elliot Murdock wrote:
Hello,
I am wondering how to send a call to a specific IP address that is different
than the host of the URI. For example, an invite to the URI is
j...@phone.com needs to be sent to the IP address
Hi all.
I'm trying to use the setvar field in my RT sip configuration to store some
information about a phone. This works perfectly for my Polycom phones. For
the most part, it works for the Grandstream as well.
However, when the GS tries to transfer a call, the variable isn't available.
2011/5/24 Kristijan Vrban vrban.l...@googlemail.com
http://code.google.com/p/zaphfc/
Hi,
Which hardware would be recommended to try this code ?
Regards
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On 20 May 2011 16:16, Ishfaq Malik i...@pack-net.co.uk wrote:
On Fri, 2011-05-20 at 10:58 -0400, Leif Madsen wrote:
On 11-05-20 09:37 AM, Ishfaq Malik wrote:
Do many people use this?
Is it reliable and safe?
It may still work, but that code is quite old, and I'm not even sure it's
Hello,
Thank you for the reply.
How would one go about adding a SIP proxy IP address in Asterisk for peers?
The host parameter for a peer (in sip.conf) seems to define both the domain
URI and the IP address for the next SIP proxy hop (there is an ipaddr
parameter, but it seems that only works
On 05/24/2011 09:56 AM, Kristijan Vrban wrote:
http://code.google.com/p/zaphfc/
Thanks! For the archives I found another site where the zaphfc code from
that google site is integrated into the DAHDI source:
http://sourceforge.net/projects/dahdi-zaphfc/
Regards,
Patrick
--
Hello,
It seems that the outboundproxy parameter in sip.conf is what is needed.
--Elliot
On Tue, May 24, 2011 at 12:15 PM, Elliot Murdock murdo...@gmail.com wrote:
Hello,
Thank you for the reply.
How would one go about adding a SIP proxy IP address in Asterisk for peers?
The host
On 05/24/2011 10:50 AM, Olivier wrote:
2011/5/24 Kristijan Vrban vrban.l...@googlemail.com
mailto:vrban.l...@googlemail.com
http://code.google.com/p/zaphfc/
Hi,
Which hardware would be recommended to try this code ?
As far as I know you can use any ISDN card with a Cologne HFC-S chip
On 05/24/2011 11:02 AM, Steve Davies wrote:
[snip]
I use astmanproxy with Asterisk 1.6.2.18 - It works fine. The most
recent version is on Github, and is not that old. In fact that reminds
me that I really must upload my latest changes to my github fork of
the project!
The last update to Dave
Hi List
With asterisk 1.4 intermittently I have an issue when i call manually using
a hard phone Snom (sip extension) a number X I found another client with
another number Y
I don’t know if I have any things wrong in my configuration .i have this
issue 1/100
Any help will be appreciated.
One of our customers has an Asterisk conference bridge connected to a
SIP trunk from an ITSP. Yesterday, they had two inbound calls that
didn't get hung up properly. From the tcpdump SIP trace that we have
running continuously, I can see that no BYE was received by the bridge,
and when some hours
24 maj 2011 kl. 12.19 skrev Tony Mountifield:
One of our customers has an Asterisk conference bridge connected to a
SIP trunk from an ITSP. Yesterday, they had two inbound calls that
didn't get hung up properly. From the tcpdump SIP trace that we have
running continuously, I can see that no
try to verifiy the zapata telephony interfaces with this command zttool
and verify the status of alarms is OK or REC
2011/5/24 James zhu zhulizh...@live.com
hi:
yes, it is true, the channel is locked by some tasks. you can please
restart asterisk and zatpel.
Best regards,
James.zhu
On 23/05/11 22:30, Elliot Murdock wrote:
Hello,
I am wondering how to send a call to a specific IP address that is
different than the host of the URI. For example, an invite to the URI
is j...@phone.com mailto:j...@phone.com needs to be sent to the IP
address 123.456.789.255, not to the IP
In article 098fafc6-cc79-49d4-bdf9-95beeb4de...@edvina.net,
Olle E. Johansson o...@edvina.net wrote:
24 maj 2011 kl. 12.19 skrev Tony Mountifield:
One of our customers has an Asterisk conference bridge connected to a
SIP trunk from an ITSP. Yesterday, they had two inbound calls that
On 24/05/11 12:08, Tony Mountifield wrote:
OK, thanks. Sounds like there was some kind of issue at the ITSP then.
I have seen this happen with broken SIP-ALGs in routers too. The ITSP
sends the BYE but for some reason a broken SIP ALG will not deliver the
packet to the right place. The
Hi all,
Just in case if anyone will be interested in BroadTel UPA-1, a USB to FXS
adapter embedded with SIP softphone. Product specification is as follows:
Hardware:
USB to RJ11 FXS adapter
1 USB port, for computer connection
1 RJ-11 FXS, for phone connection
Dimension (L x W x H): 53 x 15 x 28
On 24 May 2011 10:43, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote:
On 05/24/2011 11:02 AM, Steve Davies wrote:
[snip]
I use astmanproxy with Asterisk 1.6.2.18 - It works fine. The most
recent version is on Github, and is not that old. In fact that reminds
me that I really must upload
Hi,
I am looking for tutorial to generate a callfile so that after my program
executes, a callfile is generated and pass to asterisk to send to the
recipient.
Any suggestion?
Besides, do you know if there is a web-based GUI to send sms via asterisk?
Thanks.
CK
--
On Tue, 2011-05-24 at 10:02 +0100, Steve Davies wrote:
On 20 May 2011 16:16, Ishfaq Malik i...@pack-net.co.uk wrote:
On Fri, 2011-05-20 at 10:58 -0400, Leif Madsen wrote:
On 11-05-20 09:37 AM, Ishfaq Malik wrote:
Do many people use this?
Is it reliable and safe?
It may still work,
Hi all,
Anyone know if it's possible to force asterisk to use 'my_table' for
passwords instead of the 'secret' table?
S.
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New to Asterisk?
On Tuesday 24 May 2011, BroadTel wrote:
Hi all,
Just in case if anyone will be interested in *REDACTED*, a USB to FXS
adapter embedded with SIP softphone. Product specification is as follows:
Please refer to the fifth and sixth words of the title of this mailing list.
To everyone else, I
On 05/24/2011 11:35 AM, A J Stiles wrote:
Someone asked about the quality of it, he was quoting the hardware specs
of a similar device.
I doubt magicjack publishes that kind of detail about theirs.
so where is the problem ? Its irrelevant he represents that device
commercially.
On
On Tuesday 24 May 2011, jon pounder wrote:
On 05/24/2011 11:35 AM, A J Stiles wrote:
Someone asked about the quality of it, he was quoting the hardware specs
of a similar device.
No they didn't. The original message to which the spammer was pretending to
reply (and in the wrong place,
On 05/24/2011 11:57 AM, A J Stiles wrote:
On Tuesday 24 May 2011, jon pounder wrote:
On 05/24/2011 11:35 AM, A J Stiles wrote:
Someone asked about the quality of it, he was quoting the hardware specs
of a similar device.
No they didn't. The original message to which the spammer was
The Asterisk Development Team has announced the release of Asterisk 1.8.4.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.4.1 resolves several issues reported by the
community. Without your help this release
On 05/24/2011 11:35 AM, A J Stiles wrote:
No they didn't. The original message to which the spammer was
pretending to reply (and in the wrong place, even) never existed in the
first place.
On Fri, Mar 21, 2008 at 4:29 PM, Matthew Rubenstein...
My archives
On 11-05-24 01:21 PM, Steve Edwards wrote:
If it take the OP (of this thread) 3 years to reply, what does that say about
their product support?
Par for the course? :)
Leif.
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Steve Edwards wrote:
My archives don't go back that far
Mine do.
No match on Jack, magic or itntelecom.com
mailto:c.savinov...@itntelecom.com
Doug
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Dear;
I did restart for asterisk and actually i restarted the machine using init 6
but the problem still the same.
I run zttool and it gives OK and not REC, also when I pressed F1 for more
details I saw that
Current Allarms: No Allarms
Sync source: Internally clocked
IRQ Misses:
On May 20, 2011, at 3:30 PM, Anthony Messina wrote:
On 05/20/2011 01:20 PM, e...@erols.com wrote:
#1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to
receive faxes via T.38. Sending faxes is not a requirement. Does anyone
have a working asterisk 1.8.4 configuration
Eric
Contact me off list if you like and we can talk about your t38 fax needs.
We may be able to help depending on your location.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
From: e...@erols.com e...@erols.com
Sent: Tuesday, May
On Tue, May 24, 2011 at 12:33 PM, Doug Lytle supp...@drdos.info wrote:
Steve Edwards wrote:
My archives don't go back that far
Mine do.
No match on Jack, magic or itntelecom.com mailto:
c.savinov...@itntelecom.com
Doug
Warren Selby wrote:
A quick google for magicjack quality C. Savinovich
Interesting, it would appear that my archives are incomplete.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
On 05/24/2011 02:45 PM, Warren Selby wrote:
On Tue, May 24, 2011 at 12:33 PM, Doug Lytle supp...@drdos.info
mailto:supp...@drdos.info wrote:
Steve Edwards wrote:
My archives don't go back that far
Mine do.
No match on Jack, magic or itntelecom.com http://itntelecom.com
I know I asked this some time back, and I got no response then, and
neither did someone who asked at the start of 2009 either by the looks of
it (other than a reply from me to use a PCI card!!!)
However I now have a client who'd bought one of these boxes and installed
ISDN2e - against my
On 05/24/2011 03:43 PM, Steve Davies wrote:
[snip]
Repo updated. I have tried to merge all of the other changes that are
out there also.
Thanks Steve.
Regards,
Patrick
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On 05/24/2011 01:07 PM, e...@erols.com wrote:
I have tried faxing to the DID from 2 different fax machines connected to
different POTS lines. One fax machine is a Xerox Workcentre, and the other
is a Brother Intellifax. Can you provide some more information about your
setup? If you
Hi All,
I have a sip trunk up and running with a CUCM Express, passing calls
fine except for a comfort noise error I'm getting on Asterisk:
NOTICE[7520]: rtp.c:788 in process_rfc3389: Comfort noise support
incomplete in Asterisk (RFC 3389). Please turn off on client if
possible. Client IP:
Hi All;
To enable the compilation for the addon that is coming with Asterisk 1.8 when
doing compilation for the Asterisk, what should I do?
Regards
Bilal
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On 05/25/2011 01:20 AM, bilal ghayyad wrote:
Hi All;
To enable the compilation for the addon that is coming with Asterisk 1.8 when
doing compilation for the Asterisk, what should I do?
make menuselect and enable what you would like to be build.
Regards,
Patrick
--
On 05/24/2011 07:20 PM, bilal ghayyad wrote:
To enable the compilation for the addon that is coming with Asterisk 1.8 when
doing compilation for the Asterisk, what should I do?
https://wiki.asterisk.org/wiki/display/AST/Using+Menuselect+to+Select+Asterisk+Options
--
Jose P. Espinal
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Jose P. Espinal
Sent: Tuesday, May 24, 2011 10:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to
Dear,
do you have any successful experience for installing SHT-8C/PCI/FAX (synway)
with asterisk ?
is it compatibe with asterisk (dahdi/zaptel)?
best
--
Pezhman Lali
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On Tue, May 24, 2011 at 10:50 PM, Matt Darnell mattdarn...@gmail.com wrote:
We expect that users of Skype for Asterisk will be able to continue
using their Asterisk systems on the Skype network until at least July
26, 2013. Skype may extend this at their discretion.
It's widely believed.
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