Displaying 20 results from an estimated 700 matches similar to: "Cannot dial SIP via asterisk"
2006 Apr 12
2
How to terminate ringing call before it is answered
Is there a way to terminate a ringing call before it is answered?
I am speaking of prepaid card application in which you want to make another
call, because you current number it is not being answered, and you don't want
to hangup before dialling again.
/Obelix
2006 Apr 13
2
How to terminate ringing call before it is answered?
Is there a way to terminate a ringing call before it is answered?
I am speaking of prepaid card application in which you want to make another
call, because the current number it is not being answered, and you don't want
to hangup before dialling another number.
/Obelix
2006 Jan 17
1
Is there a key sequence to stop a call as its ringing?
Is there a key sequence to stop a call as its ringing, before the call is
answered?
The * key stops a call after it is answered, but I'd like a way to cancel the
call during the ringing phase.
/Obelix
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2006 May 21
1
Events offered by
Which Actions and events to the read/write options in manager.conf give access
to, ie the options below.
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
Are they documented somewhere?
/Obelix
2005 Jun 10
1
Wildly inaccurate CDR records
My CDR is displaying wildly inaccurate results.
When I make a call the CDR records the time between connecting into the
server and hanging up, instead of recording the time between dialling
from the server to the PSTN destination via VOIP termination.
It is alright to log the duration of the connection to the server, but
why it does not log calls for termination via voip provider is the main
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
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
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
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 Jun 06
2
Variables and status problems in AGI application
I am running a prepaid application with Asterisk. When authentication has to be
done by DTMF everything works fine. However when the user is authenticated
directly from the sip phone, the channel variables seems to disappear.
Trying to retrieve the channel status always returns -1 instead of the 6 that
happens normally. It also seems to affected the DIALSTATUS and ANSWEREDTIME
variables.
The
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 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
2011 Sep 21
3
[PATCH] Btrfs: fix orphan cleanup regression
In fixing how we deal with bad inodes, we had a regression in the orphan cleanup
code, since it expects to get a bad inode back. So fix it to deal with getting
-ESTALE back by deleting the orphan item manually and moving on. Thanks,
Reported-by: Simon Kirby <sim@hostway.ca>
Signed-off-by: Josef Bacik <josef@redhat.com>
---
fs/btrfs/inode.c | 36