Displaying 20 results from an estimated 400 matches similar to: "chan_sip.c bad file descriptor error??"
2004 Apr 28
9
chan_sip.c max number of retries?
Still getting the same error.
Apr 29 11:57:49 WARNING[1125329600]: chan_sip.c:503 retrans_pkt: Maximum retries exceeded on call 6b8b4567327b23c6643c986966334873@211.28.255.135 for seqno 102 (Critical Request)
please advise anyone!!!!!someone!!!
jai
2004 Sep 12
3
Final Help on setting up x100p
Hi.
I have installed a x100p (THE x100p for those who have seen my former
post). Now I just want to connect a "normal" phone (not an IP phone) to
the card and use it as a sip extension (I have a FWD account)... more
clearly:
I want to be able to pick up the phone and call any FWD user using my
FWD account... receive the FWD calls in that phone, and also to be able
to make normal
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]:
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",
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]
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:
2003 Jun 27
2
No dial Tone but its registering from remote site! Anyone with idea?
Hello Everyone -
Well, I think I'm getting closer with the asterisk connection. This is my
setup and I keep getting this error below in ,my /var/log/asterisk/messages
file. I have opened 5060 port on the firewall box.
I would this is Warning which I can ignore! But I see the connetcion coming
but NO DIAL TONE on mt ata186 box sitting in my 192.168.200.x site!
I'm using ATA186(cisco
2003 Apr 09
1
Asterisk dies after 3-6 hours of operation. Help!
Greetings
I have been running * for about a month now.
Configuration.
(5) Cisco 79xx IP phones
(1) XP100P
Pentium III (300mhz)
192meg memory
Redat 8.0 (updated)
It seems to run for about 3-6 hours, then the process stops. I have
noticed, that * does not stop, if I do NOT have it register to other sip
servers. (FWD and PCH).
Here is are the last few lines in the /var/log/asterisk/messages
2011 Mar 25
6
Back-to-back asterisk PRI issue
Following is my scenario to connect back to back PRI of two asterisk server. PRI cards are Sangoma A102D
[Asterisk1]------------[PRI]-Cross Cable---------[Asterisk2]
Asterisk1
; Span 1 (MASTER)
switchtype = national ; commonly referred to as NI2
context = from-pstn
group = 0,24
echocancel = yes
signalling = pri_net
channel => 1-23
Asterisk2
; Span 1
switchtype = national ; commonly
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 07
2
Asterisk : No Sound No Dial
Thanks for taking a look greg and hank. This seems to be getting bettre
everyday..help please
My sjphone is running on the same box as asterisk...i believe then the red
hat firewall should not be a problem.
Whenever i dial from CLI i get
#########
Executing Goto("OSS/dsp", "default|s|1") in new stack
-- Goto (default,s,1)
-- Executing Wait("OSS/dsp",
2003 Jul 11
1
Unable to find IP address???
This morning, I received a very strange error message on the Asterisk
console.
The error occurs when I try to access iconnect
WARNING[196621]: File chan_sip.c, Line 386 (__sip_xmit): sip_xmit of
0x80d0854 (len 649) to 213.137.73.178 returned -1: Bad file descriptor
I also get this error when I try to reload:
WARNING[16384]: File chan_sip.c, Line 5355 (reload_config): Unable to
get IP address for
2005 Oct 08
1
Cannot dial SIP via asterisk
I have been trying to connect via sip and things don't seem to work. What do
messages like this mean?
Oct 9 00:33:57 WARNING[22849]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x81ab834
(len 361) to 216.127.66.119 returned -1: Invalid argument
Oct 9 00:33:58 WARNING[22849]: chan_sip.c:694 retrans_pkt: Maximum retries
exceeded on call 000638cf3adb579455c0d20b2051ba1d@127.0.0.1 for seqno 102
2009 Nov 11
2
SIP source address error
Hi all,
My Asterisk problem today involves getting a SIP client on a private
net to register with a server somewhere else on the Internet. This
worked for me about a year ago no problem, but now I see an error
message on the remote server every time the client attempts to connect
(the server is running Debian lenny with Asterisk 1:1.4.21.2~dfsg-3).
Here's an example:
[Nov 11
2004 Jun 30
2
Remote SIP client HACK JOB
I couldn't be happier with the simplicity of this - but it's a hack!
Hi all,
I'm currently using a SIP client (BT101) to connect via DSL to a remote
instance of Asterisk.
- Asterisk has a private IP behind my OFFICE router.
- The SIP client has a private IP behind my HOME router.
I'm doing this _without_ the use of STUN or proxy servers.
Here's how it works:
-
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 Sep 25
1
Application almost there..Dialplan challenges
Aloha,
I have a Cisco Gateway the is functioning as my SIP Gateway to the PSTN.
I have an Asterisk box, RC2 with a for port FXS card providing
dialtone for a Norstar Key System.
I have it working so when you press a line key on the Norstar you get
dial tone from the Asterisk box. The user has to dial '9' then they
can dial there number which is sent to the Cisco GW via SIP and the
call
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.