Displaying 20 results from an estimated 300 matches similar to: "Asterisk dies after 3-6 hours of operation. Help!"
2003 Apr 06
1
Bug? * not correctly honouring tag on To?
Hi Mark,
Current CVS, * isn't correctly remembering the tag added to the To header
by a server.
For instance:
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.96.69.210:5062;branch=7b858c09
From: steve-ata186 <sip:asterisk@81.96.69.210:5062>;tag=14925711
To: <sip:18478974611@4.42.235.170>;tag=t2907cab0911c8g
Call-ID: 232752ec518d398f25ce03f45a8940f3@81.96.69.210
CSeq: 102 INVITE
2003 Nov 27
13
Asterisk behind NAT << How to do it.
Thanks to ww and his patch on bug #104, I have successfully implemented
Asterisk behind NAT without using STUN or anything crazy. It's quite
straight forward.
Until this gets tested enough and put into CVS, you will have to patch
your chan_sip.c file to do this. I'm sure within the next few days this
will get put merged into CVS if no one finds any problems.
I tried this on chan_sip.c
2003 Sep 23
1
App_festival crashing
Hi all,
I'm unable to put app_festival to work. I successfully patched,
installed and tested festival (interactive logon and telnet to server
port) which seems to work without problems.
But when I test it in asterisk I got the following trace in console:
-- Executing Answer("SIP/bsenicar-850b", "") in new stack
-- Executing
2003 Apr 07
1
Don't be upset !!! Architecture is need !!!
asterisk-users-request@lists.digium.com wrote:
>Send Asterisk-Users mailing list submissions to
> asterisk-users@lists.digium.com
>
>To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
>or, via email, send a message with subject or body 'help' to
> asterisk-users-request@lists.digium.com
>
>You can
2005 Oct 05
1
Attempted to delete none, xistent schedule entry 1! ??
I just upgraded my test Asterisk box to the latest CVS HEAD. "show
version" only shows "Asterisk CVS HEAD built by root....etc", with no
date or version number. I downloaded this version on Monday, Oct 3.
About once every minute, I get this while at the CLI> prompt:
sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule entry 1!
This only appeared
2003 May 02
1
WARNING (Sipsock_read) Recv error: Resource temporaily unavailable
Greetings
I am receiving following error message. Any idea as to why?
WARNING[147466]: File chan_sip.c, Line 4530 (sipsock_read): Recv error:
Resource temporarily unavailable
WARNING[147466]: File chan_sip.c, Line 4530 (sipsock_read): Recv error:
Resource temporarily unavailable
Frank...
2005 May 16
2
callback problem
hello
i am trying to make a callback solution.
client will call callback number and call is
terminated.
now callback server will create a call for that
client.
actually i have a problem in this process. that server
is creating call to client (UA) when previous call is
not disconnected yet.
UA---------->Asterisk(callbacknumber) callis answered
UA<----------Asterisk(callbackserver) call is
2004 Apr 20
1
TE410P zaptel Driver Situation
Dear List
i have upgrade my * box with the latest CVS version of Asterisk Stable 1.0
and zaptel/libpri my system is MDK9.2 with 1 TE410P and seems work well for
now but i have a little amount of traffic (25 IN/OUT calls) i only notice
this Warning.. What kind of error is?
-------------------------------
Apr 20 21:28:49 WARNING[147466]: chan_zap.c:5979 zt_pri_error: PRI: !! Got
reject for
2015 Jun 08
3
Peer unreachable after IP change
Hi list!
Another day, another problem...
I'm checking with Nagios my Asterisk at home, and since yesterday I noticed
that, after my IP changes (Deutsche Telekom drop the DSL-line every 24 hours,
so that I have a new IP every day), the peer of an VoIP-provider I use is
UNREACHABLE.
Yesterday I though it was a problem on the line, but today is the same, so I
think it is something other...
2003 Jun 08
1
Asterisk, ATA186 and callerid
Hi,
I'm having trouble getting caller*id to appear on my phone connected
to an ATA186, and being called from Asterisk.
Does anyone out there successfully see callerid on their
ata186-connected phone?
The "From:" header in the INVITE to the ATA seems to have the "right
stuff" - eg
From: "Study phone" <sip:6002@195.217.255.45:5062>;tag=as412db061
But
2004 May 28
1
* will not load, after latest CVS install
Greetings
I was getting bad static crackle on a phone, so I reload from the latest CVS and did
a make clean ; make install on zaptel, libpri and asterisk
Now I get this error
[chan_skinny.so] => (Skinny Client Control Protocol (Skinny))
May 28 13:59:42 WARNING[16384]: chan_skinny.c:2541 reload_config: Unable to get our IP address, Skinny disabled
Urgent handler
[chan_oss.so] => (OSS
2006 Nov 09
2
register suddenly fails
Hi everybody,
I've got a very strange problem:
As far as I remember I didn't change anything on my Asterisk side. I
have 2 SIP providers to which I can place outbound calls.
Today I noticed that outbound calls to provider "inode" fail (and
inbound from this provider too). On the CLI I get every 20 seconds
following messages:
Nov 9 20:01:07 NOTICE[952]: chan_sip.c:5422
2003 Sep 24
6
Festival Problems
I am trying to use festival (latest version 1.4.3)
I have downloaded all the files needed and patched it with the provided
diff.
festival does work and does tts fine.
but when I call Festival either from an extention or an AGI script, I get
this in my asterisk messages log, but no sound on the channels (H323 or SIP)
- they (the clients) just say "trying" and then hangup...
Sep 24
2003 Nov 19
1
FXO card still won't pick up...
I recently updated (fresh checkout) to the newest zaptel and Asterisk.
The one I was using before was a couple of months old.
After updating, my zap channels don't work. They won't pick up incoming
calls or dial out. When I try to dial out I get:
-- Executing Dial("SIP/3064-564c", "Zap/g1/ww954.......") in new stack
NOTICE[245776]: File app_dial.c, Line 698
2004 Mar 31
5
3-4 port FXO card recommendations
*This message was transferred with a trial version of CommuniGate(tm) Pro*
In setting up Asterisk, I'm looking to dump my current phone system (Nortel
Venture). I presently have three POTS lines.
I would use a VOIP provider, but now are presently available in the Toronto, ON,
CANADA area that support user owned hardware/software. I need a 416/647 area
code number.
In looking at FXO cards
2009 Dec 23
1
Problems with chan_sip
Calling my home numbers has always worked. Till now. The Asterisk CLI
show the following :
[Dec 23 10:53:22] NOTICE[25159]: chan_sip.c:12640 handle_response_invite: Failed to authenticate on INVITE to '<sip:092xx90xx at 85.xx.xx.xx>;tag=as5b139383'
And after restarting Asterisk, the CLI is flooded by :
[Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of
2003 May 19
6
G729 and snom
hey, I bought a license for 729 but I can't use it this is the message.
== Registered translator 'g729tolinb' from format 8 to 6, cost 99999
== Registered translator 'lintog729b' from format 6 to 8, cost 18
== Parsing '/etc/asterisk/enum.conf': Found
Asterisk Ready.
*CLI> WARNING[5126]: File chan_sip.c, Line 1601 (process_sdp): No compatible codecs!
2006 Jan 18
1
chan_sip.c:5262 sip_reg_timeout Probably a DNS error for registration
Hello,
I have a problem with an LAN-Server behind an NAT-router.
Asterisk Version 1.2.1 or 1.2.2 doesnt matter
10 minutes after starting Asterisk I loose all registrations at external
SIP-proxys.
The reason seemed to be that Asterisk send every second an request to every
sip-proxy "Request: OPTIONS sip:sip.domain.tld". Every request is responded
by the sip-proxy.
After some minutes
2008 Dec 16
1
problems of DNS
Hi list, I have for a year I have an account to call with broadvoice from
about 3 days beginning a not registered problem of, asterisk shows to a
message of error with the DNS, and my dns this working fine
WARNING[5770]: chan_sip.c:7595 transmit_register: Probably a DNS error for
registration to 908XXXXXXX at sip.broadvoice.com@sip.broadvoice.com, trying
REGISTER again (after 20 seconds)
[Dec 16
2009 Apr 26
1
1.6.1: "DNS error" but ping works
With 1.6.1 svn:
[2009-04-26 15:01:00] NOTICE[1844]: chan_sip.c:9927 sip_reg_timeout:
-- Registration for '17470121145 at proxy01.sipphone.com' timed out, trying
again (Attempt #30)
[2009-04-26 15:01:00] WARNING[1844]: acl.c:376 ast_get_ip_or_srv: Unable
to lookup 'proxy01.sipphone.com'
[2009-04-26 15:01:00] WARNING[1844]: chan_sip.c:10037 transmit_register:
Probably a DNS