2006/4/18, Doug Lytle [EMAIL PROTECTED]:
iaxmodem is a software modem that uses the IAX protocol and registers toAsterisk as an IAX client allowing HylaFAX all the resources of theAsterisk PRI or whatever allows connectivity.Thank you very much for the explaination.
1. Does using iaxmodem imply
Hi,
I had similar problem and problem was in SIP ATA device (we use Sipura
2100). They was set from factory to send 30ms voice frame,
when we change frame to 20ms everything work perfectly.
Stepan
Carlos Chavez napsal(a):
I am having a strange problem with [EMAIL PROTECTED] 2.7
Hi Steve!
I tried the gain tweaking on the E1 just to check but can you tell me if
that thing I was talking about dropping calls when gains are +2 -2 for
example is the right behavior?
I think my main problem is latency on the te10p cards. I have a te110p card
and 2 tdm04b and te110p's is
If the voice distortion sounds like clack clack clack las if you had a fan
right next to you (remember when you talk directly to a fan in front of you,
the other side gets your voice like in intervals), if thats the case,
exactly, your frame size should be 20ms, sipura and some other atas come by
Hello
I am just a newbie guy trying to find out more about T1 and pricing etc. And
I have come across a provider who starts talking about IDA-P and etc. As far
as I know it stands for Integrated Digital Access and that all I know but
would like to get a head up from some expert if possible.
Sam
What are you using as FXO ports for a few analog (remote) lines?
What is the price, where to buy, what is your experience?
bye
Ronald Wiplinger
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Err, you know i looked through that page before i posted, not sure how i
missed that one, thanks
--
~Shaun
Paul Hales [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
*${UNIQUEID}*: Current call unique identifier
- From the Asterisk Variables webpage
Linksys SPA-3000 Single Port $90
Mediatrix 1204 4 port Gateway $580
Rhino CB24 24 Port Channel Bank + Rhino R1T1 Card $2000
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original Message-
Bahh, uniq id wont work for my situation because the id for the caller and
the id for the callee are diffrent. Makes sense, they are too diffrent
calls that are going to be bridged
--
~Shaun
Shaun [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Err, you know i looked through
Hello,
Can you post your oh323.conf?
Silviu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of yusuf
Sent: mardi 18 avril 2006 17:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] correct version of asterisk for
hi i don't know if we can do that . but i guess we can use audacity .. to mix both the files and get what you want.-Giridhar BandiOn 4/18/06,
Herchi Silviu
[EMAIL PROTECTED]
wrote:
Hi all,
I'm setting up an IVR using Asterisk.
Is there a way to have two streams played to the caller
try,
database get SIP/Registry/peername
it gives you a string which contains the info, then pass it to CUT to
extract ip-adr and port
Freddi
To do that you need to get the remote ip address and port of the sip peer!
I found the function:
${SIPPEER(exten:ip)
But how can I get the port???
The problem with this solution is that the IVR uses phrases
generated on the fly using pre-recorded words and digits. So I can not pre-mix
the music, it has to be sent along.
Silviu
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giridhar
Reddy BandiSent: mercredi 19 avril
Hi
Have you tried using something like
Set(ORIGINAL_CHANNEL=from-sip)
in the original channel?
You can then use Dial(Local/number/${ORIGINAL_CHANNEL}).
Regards,
Silviu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Edwin
Groothuis
Sent: mercredi 19
Hi Daniel can you give us more information so that it would be easy to debug.like voice mail configuration etc Thanks,GIridhar Bandi.On 4/18/06,
Daniel Korndorfer [EMAIL PROTECTED] wrote:
Hi,when I call the voicemail app, it starts and die suddenly. Has anyonealready had this
Hello,
I think you should handle the fax in the h (for Hangup) extension (which is,
after your fax was received), instead of using the priorities following the fax
reception (as in your example). Have a look at the different examples in the
wiki, like
Hi,
3-way calling is implemented by the endpoint that
starts the call. I need something that I can make on
server side.
Like all of the agents in a call center has background
music while talking.
Regards,
NZR
--- Steve Totaro [EMAIL PROTECTED] wrote:
Dinesh Nair wrote:
On 04/16/06
I want to install LCDC, but I cannot find any description how to do it!
There is much in the wiki!!!
I guess one line fully understood in lcd.conf would help to get all
other lines done ;-)
voipjet rates/voipjet IAX2/[EMAIL PROTECTED]/${EXTEN} 1 1
011 announce-cost
In article [EMAIL PROTECTED],
Edwin Groothuis [EMAIL PROTECTED] wrote:
Greetings,
Somewhere on my asterisk system, a calls come in in a certain
context, for example, from-sip or from-pstn.
Then the calls gets routed through the dialplan, and a macro gets
called, and another one and then
Hm, interesting!
That is exactly what I need :-)
Thanks a million times!!
What more can you get from the database?
Is this documented somewhere?
http://www.voip-info.org/wiki/view/database+get doesn't give much info (to
say the least)
- Frederic
-Oorspronkelijk bericht-
Van: [EMAIL
This is what I do and it works well. It uses the asterisk database so
you can define which printer the fax gets printed on and/or email
address the pdf gets mailed to.
[macro-fax]
exten = s,1,Set(FAXFILE=${UNIQUEID}.tif)
exten = s,2,Set(FAXOK=no)
exten =
Hi Billy Try using safe_asterisk and see . safe_asterisk be useful if you fear asterisk may crash.--Giridhar BandiOn 4/19/06,
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
List,
The past few days the asterisk service on my server has
crashed several times. I have had it running for
ok thanks for the reply
i'll use postfix... i don't like very much sendmail :-)
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Enter the 'dmesg' command. It displays a log of kernel messages etc...
and may show up a problem.
On Wed, 2006-04-19 at 03:03, [EMAIL PROTECTED] wrote:
List,
The past few days the asterisk service on my server has crashed
several times. I have had it running for months and have made no
On Tuesday 18 April 2006 18:26, The VoIP Connection wrote:
The switch in the Budgetone is 10Base-T. If the PC NIC cannot
auto-detect or otherwise handle that, it will be a problem.
Yes! Looks like Mac mini cannot handle 10HD :) This is what I see when
BT-102 is connected to Alcatel Omnistack
Hello,
I have the following situation:
-
Someone dials in and
enters a queue;
-
Agent 1000 answers the
call using a cisco 7912 phone;
-
Agent 1000 transfers the
call using # to a external number (e.g. mobile phone);
-
The caller is now talking
directly to the mobile phone, agent
Inheritance solved the problem.. two understores infront of the var...
set(__SCREEN_FILE=/blah)
thanks.
--
~Shaun
Shaun [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Bahh, uniq id wont work for my situation because the id for the caller and
the id for the callee are diffrent.
Hi i have a Polycom Soundstation premier Basic Conference Telephone which is connected to Linksys pap2 boxi am unable to start recording using *1 . but with a normal analog phone connected to linksys pap2
i am able to start and stop recording .. i tried changing the DTMF setting but no use . can
...mhm... are you sure the agent transfers the call using asterisk transfer facility (ie, not using the transfer softkey of the phone)?
If this is the case, the agent should become free after the transfer...
2006/4/19, Mark Roeten [EMAIL PROTECTED]:
Hello,
I have the following situation:
-
Hello,
why don't you send a menu list back and the user
has to select a item and he browse to a certain url?
If not then we have to implement that.
Is that so important for you? No problem to make a firmware
version, let discuss why do you need this...
Suggestion:
SnomIPPhoneRedirect
Olivier Krief wrote:
2006/4/18, Doug Lytle [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]:
Thank you very much for the explaination.
1. Does using iaxmodem imply that, for example, incoming fax calls are
processed that way ?
- PSTN calls are terminated on TDM board (Digium, Sangoma, ...)
- the
I have a weird problem, a caller calls in, macro runs and then dumps them
into a queue. The agent then gets ringed and runs a macro. giving them
options.. Now the problem is that if another call comes in and gets placed
into the same queue it is not dispatched out to other agents logged in
Im using the asterisk
transfer facility (#). The agent dials #00612345678. The extensions.conf looks
like this:
exten = 00612345678,1,Dial(SIP/[EMAIL PROTECTED])
exten = 00612345678,2,Congestion
Any ideas?
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens picciuX
Hi all
I want to use sphinx2 with asterisk. I install
sphinx but when i type sphinx2-server, i have the
errors below:
ad_oss.c(105): Failed to open audio device(/dev/dsp):
No such device
FATAL_ERROR: server.c, line 476: ad_open() failed
Thanks for all!
Hi all,I've asterisk 1.2.5 , and what is happening is this:Sip user agent A calls a pstn phone BSip User agent Activates MOH.B starts listening.A doesn't hangup and just Disconnect Sipoftphone XLITE (exit)
B stills listenning Music on Hold and A has left Asterisk, who hangs the call? only when B
Hello,
I use ser for IM and presence and asterisk
When my sip agents send REGISTER messages I have two
records one in ser database the other in asterisk
database .
Ser manage far-end nat IM and presence (SIMPLE).
Harry
--- Douglas Garstang [EMAIL PROTECTED] a écrit
:
I don't think Asterisk
Tom Vile wrote:
I am open to suggestions as well.
On 4/18/06, Seth Remington [EMAIL PROTECTED] wrote:
Why is it a problem. I have 800 numbers through Teliax without any problems.
--
Chris Mason
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
Hi,
I have got my setup almost how I would like it now, but I have just
two last remaining issues that I cant seem to find answers too so i'd
be grateful if someone could help?
1) Since upgrading my Cisco 7960 SIP phone to P0S3-08-2-00 the phone
now displays the IP address of my asterisk server
Try asterisk -g
Regards
Josué
2006/4/19, Gareth Blades [EMAIL PROTECTED]:
Enter the 'dmesg' command. It displays a log of kernel messages etc...and may show up a problem.
On Wed, 2006-04-19 at 03:03, [EMAIL PROTECTED] wrote: List, The past few days the asterisk service on my server has crashed
Hi,
I was using Asterisk with Hylafax via IAX Modem. It works fine until I
upgraded to Asterisk 1.2.7.1
I didn't change any configuration but it seems that Asterisk does not
get the call from IAXModem anymore.
I'm doing something like this
Asterisk -- IAXModem -- Hylafax
Usually when I use
Marco Mouta wrote:
Hi all,
I've asterisk 1.2.5 , and what is happening is this:
Sip user agent A calls a pstn phone B
Sip User agent Activates MOH.
B starts listening.
A doesn't hangup and just Disconnect Sipoftphone XLITE (exit)
Sounds like an XLITE bug. On exit it should send some
Pimjai Wesnarat wrote:
Do I have to change any configuration when upgrading Asterisk?? Or
does IAXModem not work with Asterisk 1.2.7.1?
Don't know, my test setup is 1.2.4.
Doug
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Asterisk shouldn't see that the specific SIP user agent isn't there any more?On 4/19/06, Doug Lytle
[EMAIL PROTECTED] wrote:Marco Mouta wrote: Hi all, I've asterisk
1.2.5 , and what is happening is this: Sip user agent A calls a pstn phone B SipUser agent Activates MOH. B starts listening.
A
Are you sure you recompiled that module? Check your logs.On 4/19/06, Doug Lytle [EMAIL PROTECTED] wrote:
Pimjai Wesnarat wrote: Do I have to change any configuration when upgrading Asterisk?? Or
does IAXModem not work with Asterisk 1.2.7.1?Don't know, my test setup is
Marco Mouta wrote:
Asterisk shouldn't see that the specific SIP user agent isn't there
any more?
Eventually, yes. After the registration expires.
Doug
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http://www.paskambink.lt/mcc
Regards/Pagarbiai,
Mindaugas Kezys
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of [EMAIL PROTECTED]
Sent: Monday, April 17, 2006 8:13
PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Billing
Server Open Source
Here is what I did:
I downloaded the tar file of Asterisk 1.2.7.1 and complied it and
installed it using make and make install
I tested my other applications eg. call out, meet me, etc, they all work
fine.
I test the fax using sendfax -- weird, no call received at all. Also, no
IAXmodem
We are using G711 for phones to talk to Asterisk and G729 licenses at asterisk to talk to ITSP
Could you please suggest transcoder to use from G711 and G729 and which
is comptible with Asterisk. We will like to avoid using TDM if possible
Also i remember that initially we didn't have G729 and
Hello,
we are curently benchmarking an asterisk system
1034 sip users are logged into this system and the test software is
trying to establish 400 concurrent calls.
In the CLI I see the following messages:
Apr 19 14:20:51 WARNING[4045]: rtp.c:911 ast_rtcp_new: Unable to
allocate socket: Too
Hi All,
We made a VOIP application for PDA's (PALM OS) and we are using both SER
and Asterisk. SER is SIP proxy and it routes all the calls to Asterisk. On SER
we have RTPProxy also. My problem is that I am getting a weird noise or
disturbance for all the calls at an approximate time
Hi all,
with asterisk-1.2.6,
asterisk-oh323-0.7.3,
openh323-Mimas_patch2,
pwlib-Mimas_patch2,
on FC3 kernel 2.6.5-1.358,
I can make some h323 calls with no problem. However, on certain numbers,
asterisk just crashes
on the console i get:
-- H.323 call to [EMAIL PROTECTED] with codec(s)
Try lsof command in order to see what open files are attached in one
moment
Something like that will help:
lsof | wc -l (to get the count)
lsof | grep pid ( to see only the files attached to one asterisk
process)
hope it helps
Ing. Oscar Andrés Carriles
Presidente
InFoDaX Consultants
Nicolás
Correction to my original post:
My service has not been interrupted, as far as I know, and probably won't
be. But, it probably wouldn't be a bad idea to have another provider in case
Telesthetic does decide to cut NuFone off. (Although an agreement supposedly
has been reached.)
NuFone is still
What does asterisk -g do?
Im not finding anything on google.
Thanks,
William
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josué Conti
Sent: Wednesday, April 19, 2006
7:00 AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re:
dump the core i believe
-Original Message-
From: William Piper [mailto:[EMAIL PROTECTED]
Sent: Wed 4/19/2006 8:52 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc:
Subject: RE: [Asterisk-Users] Asterisk service
Below is a snipped debug on our PRI. We are getting number
only for the CallerID but the telco says they are sending us Name and Number.
We are getting the Name in a second frame but Asterisk is not passing it to the
device it rings. The message below says Presenation allowed of network
Well I know from personal experience that NuFone is working on a
solution for its customers as fast as it can. I know they found an
alternate termination provider and are working to have a solution for
the TF and Local DID's he currently has on his platform.
-Original Message-
From:
Pleaase read the archives or the wiki - you will shortly find you
need a wait in your dialplan
On Apr 19, 2006, at 8:10 AM, Jonathan k. Creasy wrote:
Below is a snipped debug on our PRI. We are getting number only for
the CallerID but the telco says they are sending us Name and
Number.
Hello William.
asterisk - g, makes with that you it initiates daemon of asterisk and it is in background.Does not forget in the CLI it to activate the command set verbose X (1-15) to monitor the events in asterisk.
I wait to have helped.
Greatings
2006/4/19, Steve Totaro [EMAIL PROTECTED]:
dump
I searched through the archives and the wiki...don't be so pissy...i
missed it I guess, my bad
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jerry Jones
Sent: Wednesday, April 19, 2006 9:17 AM
To: Asterisk Users
Chris,
I have some boxes in Japan too.
Just set it up as you set for a common T1.
INS64 is BRI. INS1500 is PRI/T1.
I never used Digium for this. GIve preference to Sangoma.
Isamar
On Tue, 18 Apr 2006, Andrew Latham wrote:
J1 is just a T1, the J2 is also very common and likely what you
-c dumps the core? useful in cases of asterisk crashing.
-Original Message-
From: Josué Conti [mailto:[EMAIL PROTECTED]
Sent: Wed 4/19/2006 9:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re:
Does anyone know how to use the Asterisk modules in external applications (C++)?
I must develop an application that encode/decode audio files from iLBC
or Speex to PCM and would like to use the Asterisk modules to do this.
But this application will not run in Asterisk box... I would like just
to
Hi,
I have got my setup almost how I would like it now, but I have just
two last remaining issues that I cant seem to find answers too so i'd
be grateful if someone could help?
1) Since upgrading my Cisco 7960 SIP phone to P0S3-08-2-00 the phone
now displays the IP address of my asterisk server
I've tested maxexpirey=120 and even with this, asterisk didn't stop the call:Scenario: SIP user agent has left without telling to asterisk it was leaving...There was a call to pstn world with MOH running...
Any tip to solve this?On 4/19/06, Doug Lytle [EMAIL PROTECTED] wrote:
Marco Mouta wrote:
Maybe this will help
http://www.voip-info.org/wiki-asterisk+sip+qualify
On Wed, 2006-04-19 at 14:51, Marco Mouta wrote:
I've tested maxexpirey=120 and even with this, asterisk didn't stop
the call:
Scenario: SIP user agent has left without telling to asterisk it was
leaving...
There was a
asterisk trunk from April 1 on fc3. Box has been up for several months
with no issues. Overnight, this remote box died, and rebooting shows the
following on the console:
exec of init (/sbin/init) Failed !!!: 20
umount /initrd/dev Failed: 2
kernel panic - not syncing: attempted to kill init
qualify=yes may overload my network .. no?On 4/19/06, Gareth Blades [EMAIL PROTECTED]
wrote:Maybe this will help
http://www.voip-info.org/wiki-asterisk+sip+qualifyOn Wed, 2006-04-19 at 14:51, Marco Mouta wrote: I've tested maxexpirey=120 and even with this, asterisk didn't stop the call:
Hi ALL,
I am trying to implement call center with no TDM components.
I started with ITSP SIP VICIDIAL/ASTERISK SIP SIP PBX
Transcoding and Recording is being done at VICIDIAL/ASTERISK
Dialer and load average is 1.5 for 12 agents and pacing of
1.1 to 1.2
Problems are being faced at dialer
Thanks for letting me know regarding the @ip-address problem. I take
it you have experienced something similar with this firmware?
The external directory is part of the cisco phones, nothing to do with
* really. The internal extensions I have set up all work fine with the
name using the
Rich Adamson wrote:
asterisk trunk from April 1 on fc3. Box has been up for several months
with no issues. Overnight, this remote box died, and rebooting shows the
following on the console:
exec of init (/sbin/init) Failed !!!: 20
umount /initrd/dev Failed: 2
kernel panic - not syncing:
Marco Mouta wrote:
I've tested maxexpirey=120 and even with this, asterisk didn't stop
the call:
Scenario: SIP user agent has left without telling to asterisk it was
leaving...
There was a call to pstn world with MOH running...
Any tip to solve this?
None.
I just confirmed this:
Dial
There are probably several messages before that one that tell you
exactly what is going on, but a hard drive is a good guess if the box
had been running fine and no significant updates have been made.
Ship out the spare!
-Original Message-
From: [EMAIL PROTECTED]
Gareth Blades wrote:
Maybe this will help
http://www.voip-info.org/wiki-asterisk+sip+qualify
My phones are already set to qualify=500
Doug
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Rich Adamson wrote:
asterisk trunk from April 1 on fc3. Box has been up for several months
with no issues. Overnight, this remote box died, and rebooting shows
the following on the console:
exec of init (/sbin/init) Failed !!!: 20
umount /initrd/dev Failed: 2
kernel panic - not syncing:
i guess you should be asking in a linux support mailing list or
something. Try not executing init as the first program, usually you
can make that by passing as argument to the kernel something like
init=/bin/bash so the first program wont be init but bash, that
would make you know if the problem
On 4/18/06, Brian Capouch [EMAIL PROTECTED] wrote:
Folks, please.
This thread has the potential to become a torrent.
It belongs on -biz as it has nothing to do with Asterisk itself.
And why all these cock robin the sky is falling statements. Nothing
has been interrupted yet and until it
How do I report a Bug to Digium? or asterisk project?On 4/19/06, Doug Lytle [EMAIL PROTECTED] wrote:
Marco Mouta wrote: I've tested maxexpirey=120 and even with this, asterisk didn't stop
the call: Scenario: SIP user agent has left without telling to asterisk it was leaving... There was a call to
Doug Lytle wrote:
Rich Adamson wrote:
asterisk trunk from April 1 on fc3. Box has been up for several months
with no issues. Overnight, this remote box died, and rebooting shows
the following on the console:
exec of init (/sbin/init) Failed !!!: 20
umount /initrd/dev Failed: 2
kernel panic -
Roger Schreiter wrote:
Alex Mosburger schrieb:
...
It is not my end hearing or producing echo. My voice is heard correctly
without any echo, but the other side hears his OWN voice several msec
...
Yes, this is, what I meant.
The other's voice is fed back by your device and running
back to the
John Novack wrote:
Eric ManxPower Wieling wrote:
John Novack wrote:
Damon Estep wrote:
There is some kind of issue with SIP transfer interaction between
some SIP phones and asterisk, I have personal experience with
Polycom phones not being able to do a blind xfer using the feature
Thanks for letting me know regarding the @ip-address problem. I take
it you have experienced something similar with this firmware?
Yup. Due to this I only run 8.2 on my phone, and use 7.4 on any other. You
can roll back to 7.4 and be ok, don't use 7.5 as it has a bug relating to
the phone
Http://bugs.digium.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marco
MoutaSent: Wednesday, April 19, 2006 10:38 AMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Music on Hold bug? User disconnect Sip user
Hi,
I am making a
Dialplan in which I have to record some outgoing calls from some users and not
from the others.
I made to outgoing
calls Macros, when that records calls (using Monitor()), and another that
doesn't: [macro-outgoinglocal] and
[macro-outgoinglocalrecord]
Is it possible to
Perpahs you need a Wait(2) before Answer() in the dialplan, because telco
send CallerIDName after some time (second ring, I suppose).
Mimmus
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jonathan k. Creasy
Sent: Wednesday, April 19, 2006 3:28
Thanks for letting me know regarding the @ip-address problem. I take
it you have experienced something similar with this firmware?
The @ip-address is actually a documented cisco fix to another problem.
I'd have to look it up, cause I don't remember exactly what it was, but
it's been on the list
On Tuesday 18 April 2006 21:06, Sean Garland wrote:
So I have * box shorewall/linux NAT firewall internet -
WRT54G with openwrt - IP500
I have 5060, 4569, and 1 through 2 forwarded to * box from
internet. I have tried everything I can think of on the wrt to get it
Did you try rtpholdtimeout in sip.conf ?
Hans
Marco Mouta schrieb:
How do I report a Bug to Digium? or asterisk project?
On 4/19/06, *Doug Lytle* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Marco Mouta wrote:
I've tested maxexpirey=120 and even with this, asterisk didn't
There are several approaches:
1. Set in Asterisk DB
RECORD/${EXTEN} on
and on outgoing calls check is set for that user than do recording
otherwise skip Monitor cmd in dialplan.
2. Set account code in sip.conf for certain user
and in dialplan you could check that variable
You can change the storage method on the Polycom phones from using
NVRAM to VRAM to increase the number of entries (limited to 25 with
NVRAM according to the Polycom Admin Guide) that a phone can store.
The relevant setting is dir.local.volatile.2meg=1 or
dir.local.volatile.4meg=1,
On Wed, 19 Apr 2006, Abhimanyu Rapria wrote:
Transcoding and Recording is being done at VICIDIAL/ASTERISK
Dialer and load average is 1.5 for 12 agents and pacing of 1.1 to
1.2
What is the average CPU utilization you observe with these load averages?
Regards,
Gerald.
On 15:08, Wed 19 Apr 06, dave wrote:
The external directory is part of the cisco phones, nothing to do with
* really. The internal extensions I have set up all work fine with the
name using the callerid=Joe Blow 1234 method you suggest. The plan
was to add customers phone numbers to the
On Wed, 2006-04-19 at 08:37 +0200, Stepan Hradsky wrote:
Hi,
I had similar problem and problem was in SIP ATA device (we use Sipura
2100). They was set from factory to send 30ms voice frame,
when we change frame to 20ms everything work perfectly.
Where in the Sipura configuration
Here is a ps aux of the services while the server is crashed. Does
anyone see any service that would have a conflict with the asterisk service?
H, maybe just me, but I personally wouldn't run anything
but asterisk on my server, yet alone adding...
cups server
font server
http
I'm using Asterisk 1.2.7.1. Has the callerid number matching functionality in
extensions.conf changed recently?
exten = ,1,NoOp(${CALLERID})
hestia*CLI
-- Executing NoOp(SIP/2944093-d24d, Cletus the Slaw Jawed Yokel
2944093) in new stack
== Auto fallthrough, channel
In sox terms is SLIN .ul (as in unsigned linear).
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
Thanks very much to you both for helping me with this issue.
On 4/19/06, Aaron Daniel [EMAIL PROTECTED] wrote:
Thanks for letting me know regarding the @ip-address problem. I take
it you have experienced something similar with this firmware?
The @ip-address is actually a documented cisco fix
Do I stop the asterisk service first and
then do asterisk g?
If I try asterisk g while the
service is running, I get:
[EMAIL PROTECTED] bpiper]# asterisk -g
Asterisk already running on
/var/run/asterisk/asterisk.ctl. Use 'asterisk -r' to connect.
[EMAIL PROTECTED] bpiper]#
Johann Steinwendtner wrote:
Did you try rtpholdtimeout in sip.conf ?
Just tried it with rtpholdtimeout=60
did a reload from the console, and tried again.
Unplugging the phone and sitting on hold for 3 minutes. Never disconnected.
Just a reminder, I'm doing this over an IAX trunk to a SIP
Carlos Chavez wrote:
On Wed, 2006-04-19 at 08:37 +0200, Stepan Hradsky wrote:
Hi,
I had similar problem and problem was in SIP ATA device (we use Sipura
2100). They was set from factory to send 30ms voice frame,
when we change frame to 20ms everything work perfectly.
Where in the
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