Hi,
I've got a test server with two IPs (one IP is virtual and moves to a backup
server if the first goes down).
I'm trying out PJSIP and specified the virtual IP for the Bind address of all
transports I'm using.
All SIP packets are now sent of the virtual IP. But the RTP UDP data is being
Hi,
Is there any reason why ODBC voicemail storage requires varchar for most
fields?
For example, is there anything stopping me using a BIGINT or similar for
origtime or INT for duration?
Kind regards,
Dan
--
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on hold, but the caller can hear the agent.
Any ideas?
Thanks
Dan Journo
Kesher Communications (UK)
www.keshercommunications.comhttp://www.keshercommunications.com
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Ø when i use the Dial the sip/105 still ringing
This should help you out
http://wiki.snom.com/FAQ/How_to_make_Asterisk_send_INVITEs_to_trigger_the_phone_for_Intercom
Dan Journo
Kesher Communications (UK)
www.keshercommunications.comhttp://www.keshercommunications.com
yes but I believe that least recent would ring one agent at a time? If my
understanding is incorrect please correct it. We are wanting to keep with
multiple phones ring to ensure coverage.
From what I've seen, I don't think this is possible. But maybe ask in the
#asterisk channel on Freenode
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: 04 December 2013 09:08
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Asterisk SIP server on windows
Hi all,
I need to build
Hi,
I'm logging cdr via odbc to mysql. It seems that there is an intermittent
problem where the CDR(userfield) isn't written to the database. The rows all
seem to be there, but that specific field is missing. The same dialplan.
Nothings changed. Probably 1 in 10 is missing the userfield.
Any
Ø I'm logging cdr via odbc to mysql. It seems that there is an intermittent
problem where the CDR(userfield) isn't written to the database. The rows all
seem to be there, but that specific field is missing. The same dialplan.
Nothings changed. Probably 1 in 10 is missing the userfield.
Ø
Ø Ok, Ive done a little more digging and it appears that some lines in CDR-CSV
are also being missed out intermittently.
Ø But the userfield in CDR-CSV is populated every time.
Ø Its Asterisk 11.5.1
Ø Ignoring ODBC for now, how do I debug writing of CDR-CSV?
I've logged this as an issue
It is really unbelievable ... I was thinking: Asterisk uses an internal
database to maintain states of peers. It is usually located in
/var/lib/asterisk/astdb and it is a berkely db, but other database backends
seem available. Are you sharing also this database between the two servers?
It
Upgrading to the latest version didn't help. After about 30 minutes,
Asterisk2 tries to send out OPTIONS keepalive packets to peers listed as
Registered on Asterisk1.
It is something really amazing... Can you run sip show peers on each one of
the servers and post the response?
You said
I am curious, is your version of asterisk correctly compiling the regserver
field? Each server needs to have a distinct server name.
Upgrading to the latest version didn't help. After about 30 minutes, Asterisk2
tries to send out OPTIONS keepalive packets to peers listed as Registered on
Hi,
We're trying to decide whether to switch back to a static file for sip.conf.
Currently we use mysql realtime but can't see any real benefit.
Why would someone choose realtime sip over static files?
Thanks
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp
functionality as realtime without the issues
described above.
Have I missed something?
Thanks
Dan
--
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com/ | Hosted
PBXhttp://www.keshercommunications.com/hostedpbx.html
T: 0161 820 8353
We have never experienced that and use realtime with multiple asterisk
servers.
We've only recently started seeing the problem.
To simplify the issue, assuming we have two servers, Asterisk1 and Asterisk2...
Asterisk1 is a primary server and Asterisk2 is a backup and used as a failover.
Maybe you are lacking some of the configuration. These is the relevant part.
rtcachefriends=yes
rtsavesysname=yes
rtupdate=yes
rtautoclear=yes
We have
rtcachefriends=yes
rtsavesysname=yes
and these we don't have but they are set to YES by default
rtupdate=yes
rtautoclear=yes
Its
Its probably an issue with the version of Asterisk we are using because I
haven't had this problem in the past.
I am running the latest 1.8 version. Which version are you running?
1.8.15.0. I'll upgrade it to 1.8.20.1 when I can and see if it makes a
difference.
--
on the
webadmin of the router.
Has anyone successfully connected a SIP endpoint (Polycom IP331) with Asterisk
on Virgin Media?
Kind regards,
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com/ | Hosted
PBXhttp://www.keshercommunications.com/hostedpbx.html
T
Dan
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com/ | Hosted
PBXhttp://www.keshercommunications.com/hostedpbx.html
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Asterisk
will pick up the phone as soon as it starts ringing.
In addition, if the caller happens to be talking before the call is answered,
BackGroundDetect(silence/10) won't work.
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com/ | Hosted
PBXhttp
Hi,
Using 1.4, I see that pickupgroup can only be between 1 and 63.
We run a hosted PBX service and need to give our client access to the call
pickup feature.
I thought that I could simply use the client's ID number for the pickupgroup
number.
The client's ID number is generated by our CRM
Why don't you just use an AGI/Table to control the pickup group? You can use
contexts or other granular features to keep any number of clients in this 63
count (IMO).
Thanks for your reply.
How would I use an AGI to handle the pickup request?
Would I control the whole pickup request? Or
My take on this is that asterisk just isn't multi-tenant friendly.
Is there any technical reason for this 63 group limit?
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Is there any technical reason for this 63 group limit?
Currently the number of callgroups/pickupgroups is limited to 63, because the
variable type that holds the group bits is a long long.
Ah, so it's a major change to get it to hold a larger number?
I was hoping I could just change the
Is there any technical reason for this 63 group limit?
Currently the number of callgroups/pickupgroups is limited to 63, because
the variable type that holds the group bits is a long long.
Ah, so it's a major change to get it to hold a larger number?
I was hoping I could just change the
Thanks Paul.
That link looks great. Is there more information available on how to apply
patches? Im not sure how patches work. Do I have to use a specific version such
as 1.4.20, or can I use any of the 1.4.x branch?
Thanks
Dan
--
/wiki/view/Asterisk+local+channels
Hope that helps. It's a little hard to explain, but try it out.
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com/ | Hosted
PBXhttp://www.keshercommunications.com/hostedpbx.html
I've got a Polycom 501 that I just can't seem to get lines 2 and 3 to work
on. Line 1 works fine.
Last time I had that issue, it resolved itself when i restarted Asterisk.
Are you able to do that?
Regards
Dan
--
_
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pbx*CLI core show function CHANNEL
-= Info about function 'CHANNEL' =-
Thanks, but i'm using 1.4. The most info I get is this:-
-= Info about function 'CHANNEL' =-
[Syntax]
CHANNEL(item)
[Synopsis]
Gets/sets various pieces of information about the channel.
[Description]
Gets/set various
Hi,
We are having a problem that is preventing users from using *2 to manage an
attended transfer.
After dialling *2, asterisk places the call on hold, but you can only dial one
digit before it times out, and the cli says:-
[2011-06-29 18:33:55] WARNING[29877]: res_features.c:938
.
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com/ | Hosted
PBXhttp://www.keshercommunications.com/hostedpbx.html
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correctly.
What method are you using to transmit the dtmf tones?
Regards
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com/ | Hosted
PBXhttp://www.keshercommunications.com/hostedpbx.html
.
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com/ | Hosted
PBXhttp://www.keshercommunications.com/hostedpbx.html
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I am at a loss.
Can you pastebin the following:-
- Run asterisk-cvvvddd and paste the output
- Pastebin your features.conf
- Pastebin your extensions.conf
I'll see if I can spot anything obvious.
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com
DYNAMIC_FEATURES=MixMonApp, either declared in your globals section of
extensions.conf, or used in a Set(DYNAMIC_FEATURES=MixMonApp) fashion on a
per channel basis in extensions.conf.
Sorry, i forgot to mention that one.
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp
declared in your globals section of
extensions.conf, or used in a Set(DYNAMIC_FEATURES=MixMonApp) fashion on a per
channel basis in extensions.conf.
Thanks to Warren Selby from http://www.selbytech.com for pointing that out.
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp
the user presses *, they are passed to the 'a' extension above and into
VoicemailMain.
I'm sure you can turn this into AGI easily enough if needed.
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com/ | Hosted
PBXhttp://www.keshercommunications.com
,nobeep)
If you don't want to record every call, you can give the operator the option of
press *1. We did this by adding the following to features.conf:-
MixMonApp = *1,self/both,Macro,mixmon
Hope that helps.
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp
Probably this will help you...
http://ofps.oreilly.com/titles/9780596517342/ch13.html#ACD_id288901
Check the section 'Controlling when to join and leave a queue'.
Thanks. Thats perfect!
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com/ | Hosted
Hi,
I'm trying to work out an issue with call queues.
I need the calls that are in a queue to be kicked out if all members are
unavailable (for example if all SIP members are having network problems).
I tried leavewhenempty = yes but that only seems works when all queue members
specifically
.
Otherwise, you need a network connection directly into the doorphone unit, and
some people don't like that because it can give a hacker/burglar, direct access
to your internal network.
Hope that helps.
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com
, and the phone can be rebooted through Asterisk to
update the settings instantly.
It also stores the phone's directory on the FTP server so users don't lose any
contacts if the phone needs replacing.
Hope that helps.
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp
Sorry, just realised I posted this to the wrong mailing list.
Dan
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the problem?
Dan Journo
Kesher Communications (UK)
Business Phone Systemshttp://www.keshercommunications.com/ | Hosted
PBXhttp://www.keshercommunications.com/hostedpbx.html
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Hi,
Has anyone had a problem setting up two registrations (on the same Asterisk
server) on one Polycom phone?
When the user tries to make a call on the 2nd line, it works fine.
But when they try the first line, the CLI says:-
Using INVITE request as basis request -
Hi,
I'm using qualify= on my asterisk server that provides outgoing pstn calls to a
few companies.
I've got one client in particular that has their own asterisk server which is
connected to my server.
This client seems to be having a nat issue. It's not a connectivity issue as
i've tried
Here's some examples:
2648061411
3190339404
I'm getting exactly the same. Odds of getting a working number, are like the
odds of winning the lottery.
My guess is they are either trying to find a voip trunk, or they are trying to
make cold calls to the extensions on my system. Sales or
When I dial into Asterisk ( I have a SIP trunk - which I need to make
sure is not faulty), I only get one-way voice communication.
The calling party, from the SIP trunk hears nothing - the extension
rings on the Asterisk server (you can see it in the CLI and hear it at
the computer), and
The adsl lines are with separate providers, so that won't work.
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I use the SG560 (http://www.snapgear.com/index.cfm?skey=1557) to do this.
Thats perfect. Any idea where they are available? I cant locate a store online.
Thanks
Dan
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Hi,
I've got a client with two ADSL connections for redundancy.
Is it possible to set up asterisk to connect to one SIP provider using both
adsl connections and load balance between the two connections?
Or to use one connection as the main one, and automatically fail over if the
first
Hi,
I've got a dialplan that transfers all outgoing calls to a Local channel before
dialling out via SIP.
I did this because sometimes i'm dialling two numbers at the same time and need
to know which call is answered for billing purposes.
However, I've just noticed that billsec is always equal
Hi,
I've got a dialplan that transfers all outgoing calls to a Local channel before
dialling out via SIP.
I did this because sometimes i'm dialling two numbers at the same time and need
to know which call is answered for billing purposes.
However, I've just noticed that billsec is always equal
Thanks Leif,
Forgot I could do a db lookup for the ddi.
Dan
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Hi,
When a VOIP user dials an external number, the calls are routed through our SIP
provider.
Is there a simple way to check whether the DDI exists locally before dialling
out to the sip provider?
Something like GotoIfExists(5551...@incoming_calls)
Currently, I'm paying for all calls,
Hi,
I'm sure this topic has been discussed before but i'm having trouble finding a
simple answer.
Whats the easiest way of sending an email from Asterisk?
I want to set up a warning so that after a Dial cmd, if the DIALSTATUS is
CHANUNAVAIL, Asterisk sends an email to the admin to check the
I recommend reviewing [1] and look for possible regressions. There
have been some update to RFC2833 over the last few months.
In 1.4.36, its a regression issue. I switched to 1.4.35 and no problems since.
Can anyone else confirm this for me?
Problems are either no dtmf tones being processed by
Hi,
Does anyone have a professional recording of someone saying Recording so I
can let the operator know that the one-touch recording has started successfully?
Thanks
Dan
[cid:image001.gif@01CB6F04.2EEFF060]
Dan Journo
IT Business Consultant
Kesher
Discussion'
Subject: Re: [asterisk-users] Recording
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Monday, October 18, 2010 2:37 PM
To: Asterisk Users Mailing List - Non-Commercial
Thanks.
Thats perfect!
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http://www.asterisk.org/hello
asterisk-users
Nope,
Its a totally normal self-built Asterisk.
Dan
Zeeshan Zakaria zisha...@gmail.com wrote:
Do you use FreePBX by any chance?
Zeeshan A Zakaria
--
www.ilovetovoip.comhttp://www.ilovetovoip.com
On 2010-10-16 6:38 PM, Dan Journo
d...@keshercommunications.commailto:d
Some service is definitely connecting to your asterisk using AMI. Such
services use username/password described in manager.conf. Usually its is some
monitoring service. Although the message says 'remote UNIX connection' but it
can be very well something from localhost. I would suggest to
I'm really struggling with this DTMF issue.
In order to test it, I've tried a few different providers and DTMF RFC2833 does
work with any of them, even though a few of them insist that it is.
Is this a bug with 1.4.36?
Has anyone else experienced this problem?
The Asterisk CLI is showing the
Whats payload used for in rfc2833?
I'm wondering if that is incompatible.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: 17 October 2010 14:54
To: Asterisk Users Mailing List - Non
Have you tried relaxdtmf or rfcc2833compensate?
Just tried it, but it didnt make a difference.
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-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: 17 October 2010 15:42
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DMTF Mode
Have you tried relaxdtmf
Hi,
Does anyone know where this is suddenly coming from?
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Hi,
Does anyone know where this is suddenly coming from?
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
Thanks
Dan
p.s.
Serious answer:
Looks like a process running asterisk -r. Do you have any sort of
AGI, cron job or perhaps a nagios check which does this?
Not so serious answer:
IT IS COMING FROM INSIDE OF THE HOUSE
Thanks for lightning my day!
Is there any way to debug this because as far as i'm aware,
Hi All,
Regarding the DTMF issue I reported where the tones werent being sent through
the provider to the pstn phones,
I ended up being told to switch to inband.
However, now, asterisk is not recognising my features (*1, etc).
Any ideas?
I've checked using tcpdump, and asterisk is still part
Hi,
Which DTMF mode do people mostly use?
I've tried SIP INFO and RFC2833 but although Asterisk recognises the tones (for
feature usage), the tones arent repeated to the end user.
So if I call a company that has a menu system, I can't use the menu.
Thanks
Dan
--
It depends upon whether you are receiving DTMF or sending, and whether you
are using a VoIP protocol or using DAHDI/Zaptel.
Sorry about the lack of info.
It's a simple SIP only setup. A handful of sip phones, an asterisk server, and
a sip provider.
The DTMF signals from the sip phones are
I just tried this:-
[test_calls]
exten = 555,1,Answer()
exten = 555,n,SendDTMF(12345)
exten = 555,n,Playback(beep)
I dialed 555 on the sip phone, nothing was heard, and then a beep...
It seems that Asterisk isn't sending DTMF. Its only able to receive.
Thanks
Dan
--
Then that likely means your phone have the correct dtmfmode, but the link
between you and the provider doesn't.
Just carried out another test to see if my provider was working properly:-
exten = INCOMINGDDI,1,Wait(1)
exten = INCOMINGDDI,n,Answer()
exten = INCOMINGDDI,n,SendDTMF(12345)
If I
Can you send us the SIP config of the sip provider (in sip.conf), removing
appropriate passwords and static IPs of course.
[provider]
type=friend
host=removed
username=removed
fromuser=removed
secret=password
context=incoming_calls
dtmfmode=rfc2833 also tried auto.
disallow=all
allow=gsm
Are you using the .gsm codec or some other flavor (ulaw, alaw, G729?)
This is from the sip.conf for the provider:
allow=gsm
allow=ulaw
This is from the sip extension:-
alaw,ulaw,gsm
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I would suggest first to make sure that asterisk is receiving DTMF fine from
your IP devices/phones. Do you have a test IVR where you can dial and press
digits and verify that asterisk is responding?
Made a quick IVR, and its working for both sides of the asterisk (between the
provider and
Based on this, your call is probably getting to the provider as ulaw (the
alaw is thrown out since it isn't in both selections; if you are in U.S. you
don't need the alaw). Try the call with higher debug (at least 5) and verify
which one is being selected.
debug 5 doesnt give me any info
How can I tell if Asterisk is sending the tones through to the provider?
You need to enable DTMF logging (logger.conf) and debug an incoming /
outgoing call.
Can you understand this? I can see the DTMF signals coming in. I pressed 5 on
the normal phone line, and then I pressed 8 on the sip
Could features.conf be preventing asterisk from repeating the DTMF tones?
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I'm on 1.4.30 and this is what I get using debug 5
-- Accepting AUTHENTICATED call from 192.168.xx.xx:
requested format = ulaw,
requested prefs = (ulaw|gsm|alaw),
actual format = gsm,
host prefs = (slin|gsm|ulaw|alaw),
priority = mine
Strange. I dont get
What is your featuredigittimeout value?
Not used. So default 1000ms.
I just added alaw to the provider's list, so now it reads (gsm, ulaw, alaw) and
its started working in a fashion.
The DTMF tones keep getting stuck. I press a number on the sip phone, and the
other party hears a tone. But
I just added alaw to the provider's list, so now it reads (gsm, ulaw, alaw)
and its started working in a fashion.
The DTMF tones keep getting stuck. I press a number on the sip phone, and
the other party hears a tone. But every few tones, it gets stuck and they
hear a long tone of about 3
Since alaw is sort of making it work, either your SIP provider or your
other party is not in U.S.
They arent in the US. Everything is in the UK.
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New
Since alaw is sort of making it work, either your SIP provider or your
other party is not in U.S.
I dont understand why the codec should make a difference if im using rfc2833.
Could you clear that up for me?
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From what I read, the codec could be trying to switch from rfc2833 to inband
during the call, causing the stuck effect.
Any way to prevent that?
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According to the WIKI, changing rfc2833 to auto in sip.conf should do the
trick.
Didnt help. I'm contacting the provider to see if they have any ideas.
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Thanks to everyone who helped me on this.
Hopefully the provider can sort out the sticking tones now.
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Since alaw is sort of making it work, either your SIP provider or your
other party is not in U.S.
Its stopped working again. This is really unusual. I didnt change anything.
I decided to do a tcpdump, and I can clearly see the rfc2833 packets being
exchanged correctly.
Why should both parties
Hi,
I'm using the applicationmap in features.conf to allow the user to press *1 and
run a macro which records using MixMonitor.
All was fine in our office (behind a nat). But when I took the sip phones to an
end user (also behind a nat), I found that *1 didnt work for outgoing calls.
But for
Hi,
I'm struggling to get the MWI set up on a few Polycom phones.
The setup is like this.
I've got a few phones in the context called [company2_phones] and I've got a
few mailboxes in the voicemail context [company2].
Therefore, for each entry in sip.conf (i'm actually using sip realtime if
I'm struggling to get the MWI set up on a few Polycom phones.
Sorted. From voip-info.
http://www.voip-info.org/wiki/view/Asterisk+RealTime
The database peers/users are not kept in memory. These are only loaded when we
have a call and then deleted, so there's no support for NAT keep-alives
Hello,
Does anyone know of a Flash or Java player that can play WAV files created by
Asterisk?
Thanks
Dan
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Does anyone know of a Flash or Java player that can play WAV files created by
Asterisk?
Found one.
http://blog.datacompboy.ru/2010/01/27/wavplayer-1-7-1-full-js-api-and-support-for-reversed-order-bits-lu-and-la/
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How do you handle replicating voice mails?
I do that by putting the voicemails into MYSQL and replicating that.
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Check out HAAST (High Availability ASTerisk) at [1]www.generationd.com
Bit out of my pricing. It must be possible to do it using downloadable
open-source.
Dan
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Hello,
Are there any guides to setting up high-availability asterisk platforms? Maybe
using Opensips.
I found this diagram, but i cant find any guides on how to go about setting it
up.
http://yfrog.com/5unetworkexampleg
Thanks
Dan
--
I checked the bug reports and all I could find was similar issues with the
Asterisk 1.6 which (according to the reports) have been resolved.
I couldnt find anyone talking about 1.4 so I created a new issue and someone
closed the case and added this note:-
This does not appear to be a bug, but
I use realtime on 1.4 and 1.6 servers but always with rtcachefriends=yes in
sip.conf
I already use that and it doesnt seem to re-register when a call comes in.
Only when the registration period expires, or the peer dials out.
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