Displaying 20 results from an estimated 4000 matches similar to: "Just getting started with Asterisk"
2004 May 26
2
SPAM MESSAGE - [Asterisk-Dev] warning message (sound card) - when I run asterisk!!!
All,
After installing asterisk on Linux, I run "asterisk
-vvvc". But I got the following warning message:
chan_oss.so] => (OSS Console Channel Driver)
May 26 00:37:58 WARNING[-1084845952]: chan_oss.c:980
load_module: XXX I don't work right with non-full
duplex sound cards XXX
== Registered channel type 'Console' (OSS Console
Channel Driver)
== Parsing
2005 Jul 03
0
no sound. "Failed to write frame"
Hi all,
Couldn't find a place to search the list archives...
I'm having issues in getting any sound using a fresh asterisk install
and a SJPhone to connect to it. I went by the instructions pointed at
the "10 minute guide", located here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+quickstart.
I installed it on a Slackware 10.1, by using no more than "make
2005 Jul 04
0
no sound. "Failed to write frame" (2nd post)
Hi all,
Couldn't find a place to search the list archives...
I'm having issues in getting any sound using a fresh asterisk install
and a SJPhone to connect to it. I went by the instructions pointed at
the "10 minute guide", located here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+quickstart.
I installed it on a Slackware 10.1, by using no more than "make
2008 Feb 14
4
domain name display issue in linux pc
Hi,
Thanks for your response on the kernel switching.I was away and could not reply immediately.
Right now, I am facing a differentissue. I have to set up DNS server using BIND on Centos 4.3. When Itype the hostname on Centos, it shows:
sipserver.vodcalocal.com
But the cli prompt has root at sipserver~ meaning only the sipserver part of the hostname is displayed. whyis this so? What is the
2004 May 21
0
unable to use EXEC in AGI
dear list
if I use EXEC in an agi script I get the following doing EXEC VoiceMailMain
-- AGI Script Executing Application: (VoiceMailMain) Options: ((null))
May 21 04:25:10 WARNING[1209214400]: chan_phone.c:422 phone_read: Error
reading:
Resource temporarily unavailable
May 21 04:25:10 WARNING[1209214400]: res_adsi.c:205
__adsi_transmit_messages: Un
able to send CAS
May 21 04:25:10
2005 Oct 09
1
Problem setting SIP incoming/outgoing
Hi
I am a newbie to * and I am having a problem which appears strange as I did
not find any mention of it anywhere in my search.
Simply speaking, I have an external SIP proxy server which I am trying to
configure for incoming and outgoing calls from my asterisk installation. So
here is my configuration in sip.conf
[general]
register =>
2004 Sep 14
2
Spawn extension.....exited non-zero
I am recieving inbound calls to my asterisk box over IAX.
And getting "spawn extension....exited non-zero" errors.
Im not entirely sure what this means, and would appreciate any help in
fixing my problem.
FYI:
********** is the inbound phone number
x.x.x.x is a remote asterisk box calling my own asterisk box.
When I choose it to dial an internal extension using this dialplan:
exten
2007 Aug 31
0
chan_sip.c:5495 sip_reg_timeout: ERROR
Hello,
I?ve been using Asterisk 1.2.18 for a while, and today, with no apparent
changes, I started receiving these messages:
Aug 31 13:26:57 NOTICE[27528]: chan_sip.c:5495 sip_reg_timeout: --
Registration for 'user at sipserver' timed out, trying again (Attempt #19)
All trunks and extensions went to:
sipserver:5060 user 120 Request Sent
011
2008 Feb 19
1
SIP Request: OPTIONS
Hi,
I have register a sip user to sip server. I can see after registration * is
sending periodic "SIP Request: OPTIONS" messages to server. but it's not
getting back any response that should be SIP 200/OK as the documents say.
3130.299707 192.168.2.113 -> 58.ab.cd.ef SIP Request: OPTIONS sip:
sipserver.net
3131.299513 192.168.2.113 -> 58.ab.cd.ef SIP Request: OPTIONS sip:
2014 Dec 12
1
Corrupt MixMonitor recordings - .gsm format
Hi all
Asterisk 1.8.11.0 on Centos 6.5
My VOIP phones are using G729 to a G729 trunk from a vendor (Centracom,
South Africa). Unlicensed G729 codec version on server.
75% of my .gsm files from MixMonitor are coming up corrupted about 3 minutes
into the recording.
The server has been up for 7 months beforehand with no problems with
recordings to .gsm format files.
I noted
2009 Nov 07
1
Trouble registering Cisco 7942
I'm trying to connect a Cisco 7942 to my Asterisk box. I have a 7960
and 7912 currently connected and functioning. I'm trying to use the
recommendations from here:
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP
I have created a "XMLDefault.cnf.xml" and it took the latest image but
the phone states it's unprovisioned? Any
2005 Sep 23
0
Problem with outbound calls
Hi everybody,
I have some problems making calls from a sip user (HT286) to the pstn trough
Digium Wildcard TE110P, i allways have an error : SIP 403
INVITE sip:0170708959@192.168.1.4;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.50.1;branch=z9hG4bK6576a5e11afe28bd
From: "test" <sip:4000@192.168.1.4;user=phone>;tag=713be5ecf76eda79
To: <sip:0170708959@192.168.1.4;user=phone>
2004 Nov 29
1
Polycom Reboot Script PRI errors!!
Kevin wrote:
> There is a reboot script posted on the wiki to reboot Polycom
> telephones. When I execute this script, I get the following messages.
> I am concerned as this is causing issues with asterisk and the PRI.
> Does anyone have any ideas why this would be happening?
>
>
>
> asterisk console:
>
> -- Remote UNIX connection
> -- Remote UNIX
2004 Oct 05
2
SIP multipart mime messages
I was messing about integration of a Cirpack softswitch with Asterisk
and banged my head against a problem previously noted on the list.
http://lists.digium.com/pipermail/asterisk-users/2003-November/026436.ht
ml
What is the status of this problem? Has it been fixed? I scrambled
through chan_sip.c, but couldn't find ay reference to "multipart".
Regards,
Jesper Dalberg
2005 May 29
0
chan_oss.c:572 oss_write: Unable to set device to input mode error
hi
i'm a newbie in asterisk...i installed asterisk but when i tried to
dial 1000 for the first time i got the following error messages and i
don't hear anything...
May 29 20:46:03 WARNING[262160]: chan_oss.c:413 soundcard_setinput:
Unable to re-open DSP device: Device or resource busy
May 29 20:46:03 WARNING[262160]: chan_oss.c:572 oss_write: Unable to
set device to input mode
May 29
2009 Aug 14
1
play prompt after hanup
Hi,
Can I play a prompt after hanging up a call? I have tried below but failed.
...
exten => s,n,Dial(SIP/1234)
...
exten => h,1,Playback(demo-instruct)
-- Executing [h at macro-safedial:2] Playback("SIP/3601-09856bf0",
"demo-instruct") in new stack
[Aug 14 17:24:03] WARNING[2496]: file.c:738 ast_readaudio_callback:
Failed to write frame
--
2008 Jul 22
1
issue with high latency
Hi,
Is there a specific latency that asterisk accepts? I encountered a
problem wherein when the latency was unusually high,my xlite's (i have 2
xlite) cannot register. but when the link suddenly went stable, the
x-lite just registered. what i forgot to look at is if the registration
packet is reaching my asterisks.
------ when xlite cannot register ---------------
Pinging
2004 May 13
0
ISDN & Voicemail: Strange Behaviour
Hi,
whenever I include a "Ringing" Line in some Voicemail Extension
I get an error when a call from the outside (via ISDN) comes in,
but it works when an internal (SIP-phone) calls the extension.
Here is my configuration for testing:
------------extensions.conf------------
[isdnext]
; strep external "101", our number and leave only extension
exten =>
2003 Oct 31
1
Problems with SIP
I'm new to Asterisk, but, Managed to get it working for outound calls from
my ATA --> Asterisk --> Cisco 2620 using SIP. However, I'm having problems
with Inbound calls from the Cisco.. Cisco 2620 --> Asterisk --> ATA .. In
fact, voice mail won't even work..
This is a snippet of what I'm getting when I try to call the ATA
-- Executing
2010 Jun 17
1
Asterisk no audio on calls problem.
Hi there,
I am trying to setup a configuration that requires me to use SIP and asterisk behind a firewall and over a VPN to a remote office and with some local Phones also.
I can't use IAX to my provider because they don't offer it and my handsets ( snom 300 ) also don't support IAX so it's all SIP.
The configuration is a follows
Asterisk PBX 10.202.17.217/24 ------>|