Displaying 20 results from an estimated 1000 matches similar to: "asterisk hung again"
2006 Jan 18
1
speex in asterisk 1.0.10
Hi,
Does anyone know how to configure speex in asterisk 1.0.10? I've
successfully installed it but cannot get any idea how to set the
quality, etc..
Thanks
Regards,
Stevanus
2006 Jan 25
1
jitterbuffer causes no sound?
Hi guys,
I 've tried asterisk 1.2.2. It work flawlessly for about 3 days then at
the third days I activated setting jitterbuffer=yes and suddenly there
is no voice when the call is picked up. It's really weird as if asterisk
stops sending rtp packet. I've checked asterisk log and found nothing
suspicious. Just weird :S.
I tried it in 3 asterisk server and all of them are having
2005 Jul 06
3
cisco 7940 + sccp issue
Hi,
Does anyone know how to make this thing (7940) work with asterisk
(chan_sccp module) ?
I've set the configuration according to the wiki and now the phone just
keep asking for CTLSEP<xxx>.tlv from my tftp server.
In the cisco's web interface, I found this in the device logs :
0x8106, 0x0, 0x12300800
0x8106, 0x0, 0x12300800
0x8106, 0x0, 0x12300800
0x8106, 0x0, 0x12300800
...
2005 Aug 19
1
sccp help
Hi,
I tried to connect cisco 7910 into asterisk system using chan_sccp.so.
But I got a major issue :
- when I called from 7910 to another sip phone in the same asterisk
server, the call took place normally.
- when I called from 7910 to another sip phone in different asterisk
server, the call is answered but I cannot hear nor say anything. The
phone just immediately lose its tone.
- when I got
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
2010 Jun 10
1
warning : sip_xmit
I'm getting a lot of these on the CLI :
[Jun 10 13:41:36] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of
0x1b381bb0 (len 554) to 192.168.12.120:2056 returned -1: Operation not
permitted
[Jun 10 13:41:37] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of
0x1b381bb0 (len 554) to 192.168.12.120:2056 returned -1: Operation not
permitted
[Jun 10 13:41:38] WARNING[4286]:
2004 May 06
0
Unable to find the source of the error: bad file descriptora
Hi,
After a few attempts, I've managed to grab the files from CVS and build it
on a rh redora box I've setup especially for Asterisk. Firstly, we're new
to the asterisk scene, so please excuse any "lame" questions which may
follow..
We're a new voiptalk.org customer. We have purchased the voip phones
(budgetone 102's) and set aside a little box to run Asterisk on.
2011 Mar 15
2
Some errors
Hello folks,
since I started with asterisk 1.8.2 I got this messages in my console when finish a call.
-- Executing [1610 at from-e1:1] Dial("SIP/xxx-00000027", "SIP/1610,60") in new stack
== Using SIP RTP CoS mark 5
-- Called 1610
-- SIP/1610-00000028 is ringing
-- SIP/1610-00000028 answered SIP/xxx-00000027
-- Locally bridging SIP/xxx-00000027 and
2004 Aug 11
0
Asterisk --> Mediatrix 1204 --> returned -1: Operation not permitted
When I try to make a call using the Mediatrix 1204 is showed on the CLI:
-- Executing SetCIDNum("SIP/2009-4df1", "1111") in new stack
-- Executing Dial("SIP/2009-4df1", "SIP/2217008@192.168.199.5") in new
stack
Aug 11 15:14:10 WARNING[1211108144]: chan_sip.c:590 __sip_xmit: sip_xmit of
0x81
40c5c (len 794) to 192.168.199.5 returned -1: Operation not
2004 Sep 16
1
Unable to dial using SIP using FWD and iConnectHere
Hi.
I cant make SIP calls from asterisk.
When I start asterisk, I get the following message: What does it means??
Asterisk is not behind NAT or Firewall.
----------------------------------
[chan_sip.so] => (Session Initiation Protocol (SIP))
== Parsing '/etc/asterisk/sip.conf': Found
Sep 16 09:52:33 WARNING[16384]: chan_sip.c:8477 reload_config: Unable to
get IP address for
2010 Jun 25
1
sip_xmit: sip_xmit returned -1: Operation not permitted
Hello,
my Asterisk CLI is flooded with the following message :
[Jun 25 21:24:57] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit
of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation
not permitted
[Jun 25 21:25:01] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit
of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation
not permitted
[Jun 25 21:25:05]
2004 Jan 08
0
Error messages during Registeration on CVS Version CVS-01/08/04-14:20:34
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Hello
I downloaded latest CVS version and got following error messages while registeration. ( CVS Version Asterisk CVS-01/08/04-14:20:34)
As a result IP Phone don't register with the Asterisk. Is it broken ?
How can I
2011 Apr 04
2
WARNING chan_sip.c:3115 __sip_xmit
Hey Guys,
Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where?
-Satish
shirley*CLI>
== Using SIP RTP CoS mark 5
-- Executing [7623 at from-sip:1] Macro("SIP/7527-00000008", "stdexten,7623,sip/7623&sip/7624") in new stack
-- Executing [s at macro-stdexten:1] Dial("SIP/7527-00000008",
2011 Feb 18
2
no progress indication
I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP
only trunks, and this server only has soft phones.
When I dial an extension and the phone is not registered, I don't get any
ring or progress indications, and eventually the Dial() times out and
drops into voicemail (as expected).
CLI output:
-- Executing [s at macro-StdExten:6] Dial("IAX2/barneveld-2036",
2011 Jun 07
1
tls/srtp: sip_xmit error: returned -2
I'm having trouble setting up tls/srtp secure communications on my
Asterisk server- I'm still rather new at working with Asterisk.
I have enabled tls and encryption and I have csipsimple with tls build
on the phone. I'm currently only testing one phone with this capability
so far, and the rest still work in the current state.
My logging looks like this with verbose turned up:
2005 Jul 18
0
chan_sip.c:939 __sip_xmit warning
Greetings,
Since the past week I've started receiving the following warnings on my
asterisk servers (FreeBSD / CVS-HEAD). This warning manifests itself
with x-lite/x-pro/eyebeam clients as well as sipura devices.
All of them have qualify=yes in their settings.
Jul 18 22:52:01 WARNING[73576]: chan_sip.c:939 __sip_xmit: sip_xmit of
0x8a3401c (len 483) to 195.x.y.28 returned -1: Address
2006 May 16
0
iax2 disconnect problem
Hi,
I'm using asterisk 1.2.7.1 and somehow my iax trunking is getting these
problem :S.
Sometimes iax acts weird and start to drop calls randomly and give these
at the log:
May 16 13:44:22 DEBUG[5264] chan_iax2.c: Immediately destroying 6,
having received INVAL
May 16 13:44:22 DEBUG[5264] chan_iax2.c: Immediately destroying 6,
having received INVAL
May 16 13:44:22 DEBUG[5264]
2004 Sep 13
0
Registering asterisk with FWD
Hi.
I have a x100p card installed and also asterisk, but I just dont get
asterisk to register with my sip provider (FWD)... when I start asterisk
using the following command I get the following messages (first, a lot
of messages show up immediatly after starting up: I'read this is normal,
then the CLI console comes out and this messages appear):
NOTICE[229390]: chan_sip.c:3922
2004 Dec 02
0
Newby with no idea
Hi folks,
thanks for your help with my last question re: japanese FXO. It doesnt sound
very compatible so I will use a SIP FXO gateway then.
Untill I find one, im just trying to get my 2 cisco SIP phones talking to my
* server. just as a learning experience for now. heres what I have so far:
2 Cisco 7960's both using DHCP and both registering with my SIP proxy server
(Brekeke OnDo on
2011 Apr 20
0
1.8.x sip error 0.0.27.191:5060 returned -1: Invalid argument
Hi All,
Again i am posting this issue because i am curious why its happening, Do you have this error with 1.8.x version please let me know.
If i dial non-exist number or peer i got following error on my CLI . even i don't have this IP address in my system this wired number coming from where i don't know. I have tried asterisk 1.8 on ubuntu, CentOS, Redhat but this error is same