Displaying 20 results from an estimated 700 matches similar to: "WARNING chan_sip.c:3115 __sip_xmit"
2011 Apr 09
1
asterisk-users Digest, Vol 81, Issue 27
I need to change the sip port from 5060 to 5061 actually we already
used 5060 for proxy to sip any idea to change 5060 to 5061 so all can
acces the sip using this port please help........................
On 4/8/11, asterisk-users-request at lists.digium.com
<asterisk-users-request at lists.digium.com> wrote:
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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 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 Apr 08
4
IAX2/0.0.29.199
Where this revers IP comes from ?
== Using SIP RTP CoS mark 5
-- Executing [7623 at from-sip:1] Macro("SIP/7527-0000006b", "stdexten,7623,SIP/7623") in new stack
-- Executing [s at macro-stdexten:1] ChanIsAvail("SIP/7527-0000006b", "SIP/7623&IAX2/7623,20,t") in new stack
-- Hungup 'IAX2/0.0.29.199:4569-5255'
-- Executing [s at
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 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
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 Apr 05
5
IAS trunk error AES encryption disabled. Install OpenSSL.
Hey Guys!
I am getting this wired error when i am calling IAX trunk. Everything works! but i want to get rid on these RED WARNING messages.. what is wrong here ? I have func_aes.so module loaded. also i remove and test but still same error.
-Satish
== Using SIP RTP CoS mark 5
-- Executing [7623 at from-sip:1] Macro("SIP/7527-0000000d", "orasebcamdial,7623") in
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
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
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
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
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
2014 Aug 13
0
WARNING[17634]: chan_sip.c:3906 __sip_xmit: sip_xmit of 0x7fe394006590 (len 609) to 83.78.150.198:60709 returned -2: Success
i'm using asterisk with tls but always get
WARNING[17634]: chan_sip.c:3906 __sip_xmit: sip_xmit of 0x7fe394006590
(len 609) to 83.78.150.198:60709 returned -2: Success
whats wrong there?
Best Regards Jakob
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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:
-