This is the problem:
two Queues
Agent logged in as agentcallback and member of the two queues.
When a call come in the queue, asterisk call the extension provided
by the agentcallbacklogin.
The need is in the extension to have a variable with the queue id.
something like:
exten =
On 5/24/06, Andy Jefferson [EMAIL PROTECTED] wrote:
Went to their site today. Site claims they are still in biz. What is
the story? What really happened to Nufone anyway?
The word dead isn't too accurate. If you pronounced dead and were
buried while in a temporary coma, you'd see that. or not
Can anybody recommend a reseller in Europe (Netherlands) for modules for the
X100P (FXO and FXS modules)?
Cost, support are important.
Also, what is a reasonable price for an X100P in Europe? Is there a difference
in price between OEM and Boxed versions?
Thanks,
Pieter
On 5/25/06, Akpome Akpoguma [EMAIL PROTECTED] wrote:
I was actually running record() application, when I pressed the # key to
interrupt the recording, it just doesnt stop
This can depend on features.conf, the codec used, the phone used, the
digitmap of the phone if there is one and several
Hi Guys,
I'm having sound problems when diverting a call using [EMAIL PROTECTED] 1.5. I
am using the following configuration in extensions_custom.conf,
extensions_additional.conf and extensions.conf
[custom-Sales]
exten = s,1,SetVar(DivertNumber=02)
exten = s,2,Dial(SIP/116, 15)
Certainly not since it's not working properly yet.
Kevin P. Fleming wrote:
Erick Perez wrote:
does anybody knows if this patch made it into Asterisk Business Edition?
http://bugs.digium.com/view.php?id=4825
ABE never includes any features that are not in open source Asterisk,
except
Changing firmware revs did not help, so that left the LAN.
I looked long and hard at the LAN and it was basically narrowed down to the
switches. In this smaller install, several cheapo Dlink ($30) switches
de-aggregate a Cisco Catalyst switch. What I noticed was that any phone
plugged direcly
I looked long and hard at the LAN and it was basically narrowed down to
the
switches. In this smaller install, several cheapo Dlink ($30) switches
de-aggregate a Cisco Catalyst switch. What I noticed was that any phone
plugged direcly into the Catalyst did *not* lock up or reboot. Any phone
Hi All,
I have registered abhijit for SIP in asterisk Server.
I am able to register my softphone (SJPhone) to the server using the
name abhijit.
But whenever I try to make any calls I am gettinh the following error
message:-
*CLI
-- Registered SIP 'abhijit' at 172.20.28.85 port 5060 expires
On Fri, 2006-05-26 at 10:11 +0200, Remco Barende wrote:
Thanks for your input!
Previously I was using Nortel 10/100 switches, I replaced them some
weeks ago with 3C16479 gbit switches. The phones are connected directly to
the gbit switches. By coincidence I dit notice on one phone that in
Hello,
I have problem to make SIP calls, i have asterisk +
PC InterP4 + Digium TDM400P
here is the content of the sip.conf:
[SIP_PROVIDER]
type=peer
fromuser=testcomclient
username=testcomclient
secret=testr
host=IP_SIP_PROVIDER
;dtmfmode=rfc2833
context=interne
canreinvite=no
;allerid=Beer
Hi
Im very new to Asterisk and this is my first
posting to this mailing list. I got a [EMAIL PROTECTED] V2.8 working,
and now Im trying to use Asterisk.NET (http://sourceforge.net/projects/asterisk-dotnet)
to get call events to my C# program.
Asterisk.NET comes with a sample program
On Fri, 26 May 2006, Dave Cotton wrote:
On Fri, 2006-05-26 at 10:11 +0200, Remco Barende wrote:
Thanks for your input!
Previously I was using Nortel 10/100 switches, I replaced them some
weeks ago with 3C16479 gbit switches. The phones are connected directly to
the gbit switches. By
My guess would be to check your manager.conf[admin]secret = amp111deny=0.0.0.0/0.0.0.0permit=10.0.0.1/255.255.255.0read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,userThe line permit=10.0.0.1/255.255.255.0 should be adjust to your network
Hi All,
Having a rather annoying problem with the Polycom 301 phones, suspect it
to be my dialplan.
Basically if you lift the receiver off the handset and dial a number, it
will not let you dial a number longer than 10 digits (Can see this being
acceptable in US, but in UK its a right pain).
>From what I understood Zaptel was ported to the Mac quite some time ago.
http://lists.digium.com/pipermail/asterisk-users/2004-October/060872.html
Also, TerraSoft sponsored an Asterisk port to YellowDog Linux on PPC - from what I gather, with full Zaptel support.
Could you just set the variable in the part of the dialplan where they enter the queue and then reference it here?Thanks,KyleOn 5/26/06, Massimo Nuvoli
[EMAIL PROTECTED] wrote:This is the problem:
two QueuesAgent logged in as agentcallback and member of the two queues.When a call come in the
Hi all I want to install a TDM400P card. I use fedora core 4 and the version of my kernel is 2.6.11-1.1369_FC4smp. When i type lspci, i have this message: 02:01.0 Network controller: Unknown device e159:0001 how can i fix this problem? Thanks for your help! Serge MESSA OVONO
Yahoo! Mail
Is it possible to run a command in the voicemail.conf
file to change the from email-address. This way the user who gets
the email, can reply on the mail just by clicking answer. I want
to do something like this
serveremail='grep
${VM_CIDNUM}) /etc/asterisk/voicemail.conf | cut -d, -f3'
Hi Jan,
maybe externnotify voicemail.conf command may help you to exec an
external script.
Giorgio Incantalupo
Jan Pringels wrote:
Is it possible to run a command in the voicemail.conf file to change
the ‘from’ email-address. This way the user who gets the email, can
reply on the mail
On 5/26/06, El Flynn [EMAIL PROTECTED] wrote:
Hello,
Does anyone know the maximum number of queues that can be defined in an Asterisk
system?
Queues and their members are both stored as linked lists in
Asterisk's memory so there really isn't a technical upper limit in
the amount you can
I have two HFC ISDN Cards, configured using mISDN on asterisk svn head 1.2
These two cards are connected to 2 ISDN Lines, receiving calls for 50
numbers.
Everything is OK on 75 % and bad on 25 %
When is bad, In /var/log/asterisk/full I see
May 26 09:55:28 WARNING[24410] chan_misdn.c: Extension
Jamie Heckford wrote:
Hi All,
Having a rather annoying problem with the Polycom 301 phones, suspect it
to be my dialplan.
This would be incorrect.
Basically if you lift the receiver off the handset and dial a number, it
will not let you dial a number longer than 10 digits (Can see
On 26/05/2006, at 7:49 PM, Jamie Heckford wrote:
Can anyone shed any light on this issue? I thought it could be
asterisk
is trying to Dial to soon so I added a Wait in the dialplan but it
didn't seem to work.
Polycoms have their own dialplan built into the phone. Depending on
how you
Marco youre
advise worked like a charm! I put in the IP of my PC and now the
authentication works and I can see all events. Thank you very much. J
Werner
Message: 8
Date: Fri, 26 May 2006
10:35:00 +0100
From: Marco
Mouta [EMAIL PROTECTED]
Subject: Re:
[Asterisk-Users]
Kyle Sexton wrote:
Could you just set the variable in the part of the dialplan where they
enter
the queue and then reference it here?
Yes, that is the way to do this. Set a variable in the dialplan before
putting the _caller_ into the queue, and prefix the variable with at
least one underscore
serge messa wrote:
I want to install a TDM400P card. I use fedora core 4 and the version of my
kernel is 2.6.11-1.1369_FC4smp. When i type lspci, i have this message:
02:01.0 Network controller: Unknown device e159:0001
how can i fix this problem?
There is no problem. The TDM400P
Kyle Sexton ha scritto:
Could you just set the variable in the part of the dialplan where they
enter the queue and then reference it here?
:-) very simple, tested but not working, and logically i think it is
right.
In asterisk a variable (dialplan SET) is bound to the incoming
channel, but,
Kevin P. Fleming ha scritto:
Kyle Sexton wrote:
Could you just set the variable in the part of the dialplan where they
enter
the queue and then reference it here?
Yes, that is the way to do this. Set a variable in the dialplan before
putting the _caller_ into the queue, and prefix the
I looked long and hard at the LAN and it was basically narrowed down to the
switches. In this smaller install, several cheapo Dlink ($30) switches
de-aggregate a Cisco Catalyst switch. What I noticed was that any phone
plugged direcly into the Catalyst did *not* lock up or reboot. Any phone
Hi,
during gradual migration to Asterisk, I put Asterisk in front of a legacy
Alcatel PBX:
PRI PSTN -- Asterisk -- E1 cable -- Alcatel PBX
After successful deployment of VoIP phones, it's time to drop Alcatel PBX!
I'd like to keep some of analog lines to support modem, fax and some older
stuff.
Remco Barende wrote:
On Fri, 26 May 2006, Dave Cotton wrote:
On Fri, 2006-05-26 at 10:11 +0200, Remco Barende wrote:
Thanks for your input!
Previously I was using Nortel 10/100 switches, I replaced them some
weeks ago with 3C16479 gbit switches. The phones are connected
directly to
the
I just installed [EMAIL PROTECTED] 2.8 SW on a DELL box. I can connect from my webbrowser to the AMP GUI and can with no problem work with it.
The DELL box has 2 NICs and is connected itself to an ADSL router. The
firewall is on the external NIC (eth0), the firewall of the router is
switched
Massimo Nuvoli ha scritto:
Kevin P. Fleming ha scritto:
Kyle Sexton wrote:
Could you just set the variable in the part of the dialplan where they
enter
the queue and then reference it here?
Yes, that is the way to do this. Set a variable in the dialplan before
putting the _caller_ into the
Hello to all,
Im trying to use DeadAGI to implement billing with Asterisk2Billing.
Before the billing, I had something like:
exten = _2,1,Dial(SIP/[EMAIL PROTECTED])
Now, with Asterisk2Billing would be something like this?
exten = _2,1,Answer
exten = _2,2,Wait,2
exten
I would like to suggest using any managed switch and hard setting the
ports to 100/full
I have found that the auto negotiation algorithm is generally to
blame on many switches.
As an example, connecting a cisco router to a netgear/dlink/3com/etc
will geneerate errors on the cisco
Blaming the 3com switch is very likely to be the wrong root cause. High
probability the 3com was not configured properly for the phone.
Just curious - what configuration issues did you have in mind?
- Mike
___
--Bandwidth and Colocation provided by
Can someone shed some light on why the hint feature
was implemented in the priority field that is purely an integer
in the rest of the dialplan?
There seems to be a conflict with realtime and the hint
priority, in order to put in the hints you would have to change the priority
column in
Dr. Michael J. Chudobiak wrote:
I looked long and hard at the LAN and it was basically narrowed down
to the
switches. In this smaller install, several cheapo Dlink ($30) switches
de-aggregate a Cisco Catalyst switch. What I noticed was that any phone
plugged direcly into the Catalyst did *not*
Dr. Michael J. Chudobiak wrote:
Blaming the 3com switch is very likely to be the wrong root cause.
High probability the 3com was not configured properly for the phone.
Just curious - what configuration issues did you have in mind?
- Mike
Replacing it with a Catalyst?
Andrew
Hi, folks.
I want to buy FXS-SIP terminal with 4 ports (up to 250$).
Do you have any recomendations and Asterisk configurations samples for
such devices. Any pitfalls? Actually i realy don't know what to buy?
--
=
= Best
Hi Friends, At present, I am using VoIPJET.COM provider for make calls to USA. I have two doubts. 1) I am unable to make call to UK Mobile phone. Why? 2) I want to make calls to "Turkey" country from "India". With VoIPJET, I am unable to make call to "Turkey" and unable to find VoIP provider
Plainvoip has a very good A-Z and I have
found they are fairly inexpensive.
They also offer TollFree orig and some
local dids.
www.plainvoip.com
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy
Sent: Friday, May 26, 2006 9:21 AM
To:
Dr. Michael J. Chudobiak wrote:
Blaming the 3com switch is very likely to be the wrong root cause.
High probability the 3com was not configured properly for the phone.
Just curious - what configuration issues did you have in mind?
A partial list of issues that we've seen in the last 12 years
Andrew D Kirch wrote:
Dr. Michael J. Chudobiak wrote:
Blaming the 3com switch is very likely to be the wrong root cause.
High probability the 3com was not configured properly for the phone.
Just curious - what configuration issues did you have in mind?
- Mike
Replacing it with a Catalyst?
More cowbell?
Shit, you owe me a new keyboard! Funniest thing I've *ever* read on the
list.
I've experienced the auto-negotiate issue with Snom's before. I forgot to
mention that we make it part of our standard install to force 100baseT-full.
I've also noticed the Catalyst does the
Sean Cook wrote:
Rob Lith wrote:
Does the sangoma handle sharing interrupts in some other way?
from: http://www.voip-info.org/wiki/view/Sangoma
There are no known compatibility issues (IRQ, IO etc) with ANY Sangoma
hardware and ANY make/brand of PC/server- NONE
The pci bus was designed to
On Fri, 26 May 2006, Rich Adamson wrote:
You mean that 3Com switches are not to be regarded as decent switches? At
least Snom could have put some remark then that you need a certain brand of
switches. If 3Com is not good enough for the phones I would have bought
different phones.
Blaming
>From my point of view, using cheap or expensive switch is not the point.The point is what kind of switch implementation Snom phones require ?.Up to now, it seems that problems relate to auto-negociation.
Would it be possible for anyone to check that, comparing fixed and auto-negociated behaviours
2006/5/26, Mimmus [EMAIL PROTECTED]:
Hi,during gradual migration to Asterisk, I put Asterisk in front of a legacyAlcatel PBX:PRI PSTN -- Asterisk -- E1 cable -- Alcatel PBXAfter successful deployment of VoIP phones, it's time to drop Alcatel PBX!
I'd like to keep some of analog lines to support
No you can't. We've actually got a patch that does exactly that. It uses
realtime heavily, mainly to pull the information directly from the
database. If you're interested, let me know :)
On Fri, 26 May 2006, Jan Pringels wrote:
Is it possible to run a command in the voicemail.conf file to
Remco Barende wrote:
On Fri, 26 May 2006, Rich Adamson wrote:
You mean that 3Com switches are not to be regarded as decent
switches? At least Snom could have put some remark then that you need
a certain brand of switches. If 3Com is not good enough for the
phones I would have bought
(I hope this isn't html - Thunderbird is so annoying)
I'm new to using hints/subscriptions on * so please be patient with me.
I have two * systems in different geographic locations, connected via IAX
Location1 has a Polycom 600 and a GXP-2000 phone
Location 2 has a single GXP-2000.
With the
Colin Anderson wrote:
More cowbell?
Shit, you owe me a new keyboard! Funniest thing I've *ever* read on the
list.
I've experienced the auto-negotiate issue with Snom's before. I forgot to
mention that we make it part of our standard install to force 100baseT-full.
I've also noticed the
On Thursday 25 May 2006 16:11, Sean Cook wrote:
What could be the other causes? I have exhausted everything I know how
to do. PCI sharing explains it (whether or not it is infact the
problem). This card shares the BIOS assigned interrupt with the network
card...
Audio problems can come for
First question, is there a forum for [EMAIL PROTECTED] specific questions?
I've asked what must have been questions about [EMAIL PROTECTED] here and
gotten some indication they weren't welcome.
Second, does anyone know what files need to be backed up? I don't need
to back up the entire
This is whatam doing for voicemail during my
transition.
My pbx send the 9 out on all calls. (made my
asterisk configs easier.)
All of my extensions start with 5. asterisk
extension are 56XX and 57XX, Legacy extensions are 51XX and 52XX.
I added the below lines to my
dialplan.
exten =
On Thursday 25 May 2006 17:48, mustardman29 wrote:
Just remember that USB audio devices such as a USB headset increases CPU
usage compared to standard audio. It's probably not much of a problem for
modern processors but I don't have any direct experience with them to
confirm this.
Not
Hello mates, im having calls of about 120 o 130 minutes in my accounting DB but these are calls not made by users.I guess my asterisk is not catching some BYE requests and after some timeout it hangs up the call.Is this issue known? Is there a way to trace this problem?
Im using Asterisk 1.0.10 on
[EMAIL PROTECTED] is welcome as long as you are referring to the asterisk
portion of it
and not the GUI or dialplans that make [EMAIL PROTECTED] different from the
typical
asterisk.
I believe [EMAIL PROTECTED] offers a backup button that will backup all
pertinent files
for you. I.e. dialplan,
Jim,There are SourceForge.net forums for [EMAIL PROTECTED] where you'll probably find better answers to your AAH questions. They are located here: https://sourceforge.net/forum/?group_id=123387
In terms of backup, AAH has a built-in backup feature as part of the FreePBX GUI. Set your schedule,
I also had this same problem with an older version of
asterisk. The issue disappeared when I upgraded.
Upgrade to a newer version see if it still exists.
bp
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Francisco Seratti
Sent: Friday, May 26, 2006 10:51
AM
Andrew Kohlsmith wrote:
On Thursday 25 May 2006 16:11, Sean Cook wrote:
What could be the other causes? I have exhausted everything I know how
to do. PCI sharing explains it (whether or not it is infact the
problem). This card shares the BIOS assigned interrupt with the network
card...
You might try these sites:
http://sourceforge.net/forum/forum.php?forum_id=420324 Backup has
been discussed many times here. Unfortunately, the SF forums suck in
terms of searching.
http://www.freepbx.org/
http://aussievoip.com.au/wiki/index.php?page=FreePBX
I'd like to keep some of analog lines to support modem, fax and some older
stuff. What's the best choice? A channel bank or a TDM2400P card?
Can I use a TDM2400P board together with the actual TE410P?
As far as I know, Digium doesn't support FAX through TDM2400P, even
less a modem call.
I had to
Fernando Lujan wrote:
Steve Underwood wrote:
The problem is almost solved. The card was configured as a T1 interface,
the selled came and jumped it.
Now I have the following problem. When I call from my legacy pbx, appear
a event:
*CLI May 26 12:04:09 WARNING[5215]: chan_unicall.c:627
Hi all,
I'm having trouble with incoming IAX2 calls on 1.2.7.1 - mostly they
work, but sometimes the caller just gets dead air or disconnects. IAX2
debugs show HANGUP and INVALID codes in these cases, rather than a
proper RINGING transaction.
My firewall is doing NAT, and changing the
http://www.wirelessiq.com/content/newsfeed/7319.html
Im surprised, I thought DECT was already available in
the USA from my days selling this at Ericsson Australia back in 1995.
Can someone confirm that they arent already
available?
Cheers,
Dean
Dr. Michael J. Chudobiak wrote:
Hi all,
I'm having trouble with incoming IAX2 calls on 1.2.7.1 - mostly they
work, but sometimes the caller just gets dead air or disconnects. IAX2
debugs show HANGUP and INVALID codes in these cases, rather than a
proper RINGING transaction.
My firewall is
Olivier Krief [EMAIL PROTECTED] wrote:
PS: How many users were at start connected to Alcatel PBX ?
What did you do for voicemail during migration?
I had ~110 extensions.
During migration, I simply avoid to give Asterisk goodies to Alcatel users.
Every extension migrated to VoIP could be
Nuthin beats an Atlas:
http://www.adtran.com/adtranpx/Doc/0/TUA2HMOPDK3KN6S9LM1FH91169/61200305L2-8
A.pdf
Telephony Swiss army knife. You can make it do anything. Be prepared to crap
your pants when you see the price, though.
-Original Message-
From: Time Bandit [mailto:[EMAIL
We attended the Astricon in California, US last year. Although it was not
what we expected, we did feel like we gained enough knowledge to make it
worth the time and expense to attend.
Good luck and let us know how you like the show if you end up attending.
--Todd
--
VoIP Street
DID
On Friday 26 May 2006 11:15, Rich Adamson wrote:
Have you dug into the TDM400 far enough to know whether the common
complaints are associated with a hardware design issue, TigerJet issue,
or driver? (eg, can any of the issues truly be addressed?)
My personal opinion is that the TJ320 (the PCI
Hi
Just moved offices in the UK and moved our Asterisk box from old one to new
one. Using idefisk softphones, Junghanns quadbri card for ISDN 2e interfaces.
At both offices we had one standard number and a DDI range, routed with
Asterisk.
We'd set up the configuration so each idefisk set its
Thanks, will try this... I actually don't really want to delay incoming
calls before the attendant, but it seems to take about 7-10 seconds from the
time I dial until the AA picks up, without a ring, it just sounds odd, like
the call didn't go through...so I wanted to experiment with trying to add
We had a problem like this until BT enabled callerid (an optional
extra) on the line.
Julian.
Paul Redstone wrote:
Hi
Just moved offices in the UK and moved our Asterisk box from old one to new
one. Using idefisk softphones, Junghanns quadbri card for ISDN 2e interfaces.
At both offices we
On Fri, May 26, 2006 at 04:58:34PM +0100, Paul Redstone wrote:
[snip]
I found a wiki which said that the DDI numbers we want as caller IDs need to
be
flagged as allowed CallerID number - this is done by BT - but BT do not seem
to
understand this.
Also our old local exchange was a System
Hi Jamie,
Take a look at the dialstring in your sip.cfg - you'll need to adjust
this to match your local dialing plan.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On 26-May-06, at 2:49 AM,
Nuthin beats an Atlas:
http://www.adtran.com/adtranpx/Doc/0/TUA2HMOPDK3KN6S9LM1FH91169/61200305L2-8
A.pdf
Telephony Swiss army knife. You can make it do anything. Be prepared to crap
your pants when you see the price, though.
At that price, I'll keep my dedicated analog line.
but thanks for
On Fri, 26 May 2006, Steve Kennedy wrote:
Err System Y ? System X is a Marconi switch, I didn't think they made a
Y variant, but hey maybe they do.
System Y is/was a common synonym for AXE10.
___
--Bandwidth and Colocation provided by Easynews.com --
Andrew Kohlsmith wrote:
On Friday 26 May 2006 11:15, Rich Adamson wrote:
Have you dug into the TDM400 far enough to know whether the common
complaints are associated with a hardware design issue, TigerJet issue,
or driver? (eg, can any of the issues truly be addressed?)
My personal opinion
There is a system Y, believe it or not it was introduced after system X
BT exchange classes:
TXS Strowger
TXK Crossbar
TXE Electronic
TXD Digital further sub categorized as System X or System Y
System X - GEC Plessey Telecommunications (GPT)
have you tried to set the priority to -1 for the hints in the db?don't know if it works, but I saw somewhere that the 'hint' priority was actually -1 inside asterisk... maybe this will work with realtime arch.Just a suggestion...
2006/5/26, Damon Estep [EMAIL PROTECTED]:
Can someone
Hey everyone,
A few employees have noticed some problem here and there when trying to
make outgoing phone calls. After it happens, they try again, and are
able to call through.
The dial plan for outbound calling looks like below. Which I know they
are getting to the Congestion part (which
If memory serves me properly what you are showing looks correct. You
server is registering to your provider on port 4569 as it should. Their
server is seeing you register from 64.26.155.62 and using the prt 14353
which is the port that your firewall has given that outgoing connection.
Hi the list !
I share Ethernet card IRQ with my TDM2400 without any trouble here, on an
old Intel motherboard and an old PII400 !
This is another proof that sharing IRQ is not necessary an issue.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
On 5/26/06, Mimmus [EMAIL PROTECTED] wrote:
Hi,
during gradual migration to Asterisk, I put Asterisk in front of a legacy
Alcatel PBX:
PRI PSTN -- Asterisk -- E1 cable -- Alcatel PBX
After successful deployment of VoIP phones, it's time to drop Alcatel PBX!
I'd like to keep some of analog
Hi Abhijit,
The error message says it all really. You have a context called
'internal' but not 'from-internal'
With thanks,
Tim
- Original Message -
From: Abhijit [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, May 26, 2006 9:46 AM
Subject:
Hi,
I know that this is not the right place, but Im not aware of any alternative.
I have some VOIP equipments I would like to trade-up
Digium IAXyS101i
Used for 2 days
VoipSupply page: http://www.voipsupply.com/product_info.php?products_id=772
VoipSupply Price: $90
Linksys WBP54G 802.11G WIFI
On 25 May 2006, at 20:43, Dr. Michael J. Chudobiak wrote:
I've been having problems with incoming IAX2 calls - some work, but
a large fraction are answered with dead air or disconnects from
my IAX provider.
Disabling the jitterbuffer seems to eliminate the problem (so far)!
Has anyone
On 5/26/06, Dan Elder [EMAIL PROTECTED] wrote:
Thanks, will try this... I actually don't really want to delay incoming
calls before the attendant, but it seems to take about 7-10 seconds from the
time I dial until the AA picks up, without a ring, it just sounds odd, like
the call didn't go
On 5/26/06, Francisco Seratti [EMAIL PROTECTED] wrote:
Hello mates, im having calls of about 120 o 130 minutes in my accounting DB
but these are calls not made by users.
I guess my asterisk is not catching some BYE requests and after some timeout
it hangs up the call.
Is this issue known? Is
I've been using Asterisk 1.2.6 with a 4 port FXO Sangoma A200 card for
the past month without many problems (other than the fact that the
Sangoma card doesn't disconnect hung up calls immediately, which I
posted about in another thread, and has still not been fixed),
however, I had a call from
Hi Josue, benchevWith your guidance, I want to get back to HiPath right now. But I am on the road, so I can get in touch with that system only on Tuesday. But that's really great new Josue, that you can work out the things from those commercial system.
I will be back very soon,Thanks
There isn't quite enough info in that log to tell what is going on.
What you have above is part of 2 separate conversations.
You have the tail end of a successful registration with 70.87.18.51
and the HANGUP of a call with 64.26.157.230 which your asterisk seems
to be confused about.
Could you
Mike Garey wrote:
It turns out that the Sangoma card had suddently decided to stop
answering on channels 2,3 and 4, so if someone was using channel 1,
then no other calls would be picked up. We could, however, make
outgoing calls. I tried restarting Asterisk and it didn't make a
difference.
Well NAT isn't the problem here. I just plugged it directly into the
Internet, and it still has the same problems, any other ideas.
William Piper wrote:
Doesn't the Pap2 have a setting for stun? If so, try that set it to
stun.fwdnet.net, at least for testing. If you need to use it for a ton
I put my hints in a separate static context, then set the
subscribecontext in sip.conf to make subscriptions look at that context
for hints. Perhaps that would work for you?
-Jason
Damon Estep wrote:
Can someone shed some light on why the ‘hint’ feature was implemented
in the ‘priority’
Stuart Elvish - Dallas Delta Corporation Pty Ltd wrote:
Just come across a problem - we have sent out heaps of PAP-2 ATA's and
just discovered that when joined in a conference they are choppy on
the up leg (so the other users in the conference will hear them with a
choppy sound) but the down
you can specify the logging level in unicall.conf
logleve=0-255
select a value from 0 to 8. According to unicall.h the levels are:
enum
{
UC_LOG_ERROR= 1,
UC_LOG_WARNING = 2,
UC_LOG_PROTOCOL_ERROR = 3,
UC_LOG_PROTOCOL_WARNING =
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