Displaying 20 results from an estimated 1000 matches similar to: "Problems with chan_sip"
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
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
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
2010 Aug 02
1
SIP Status: 401 Unauthorized (0 bindings)
Hi,
I have made a fresh install of asterisk-1.6.2.10 and when I register my
soft phone it gives following error. Rest are default configurations.
32.454370 MY_IP -> ASTERISK_SERVER_IP SIP Request: REGISTER
sip:ASTERISK_SERVER_IP
32.454505 67.19.43.202 -> MY_IP SIP Status: 401 Unauthorized (0
bindings)
36.454814 MY_IP -> ASTERISK_SERVER_IP SIP Request: REGISTER
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
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.
2007 Mar 23
1
upload progress bar don''t work...please help
Hello,
I''m trying to install the upload_progress gem and i can''t see the upload
progress status.
My config is :
apache2 with mod_proxy
mongrel
rails
upload.rb : ##############
require ''rubygems''
require ''drb''
require ''gem_plugin''
GemPlugin::Manager.instance.load ''mongrel'' => GemPlugin::INCLUDE
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
2004 Apr 28
2
chan_sip.c bad file descriptor error??
hi
new user here
cant seem to get fwd running, got asterisk from download site as tarball, did the readln and openssl start. Also configured the sip.conf and extensions.conf but an error with the chan_sip.c shows up?
any ideas...somebody...anybody!
thanx
jai
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
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
2013 Nov 18
3
Is it possible to evaluate a string as a parameter name?
Hi,
I''m looking to combine a couple of fact names with the value of a class
parameter to create and lookup the resulting fact''s value. Is that
possible? For example, my class will take the parameter "my_default_nic"
from beyond. So I know that as long as $my_default_nic exists on the
client, then so will facts like macaddress_<NIC>, netmask_<NIC>,