Displaying 20 results from an estimated 30000 matches similar to: "Incoming call: Got SIP response 503 "Server error" back from xxx.xxx.xxx.xxx"
2006 May 30
1
Got SIP response 405 "Method not acceptable" back from xxx.xxx.xxx.xxx
Hi, Im trying to register to a SIP provider that told me that they
only need to authenticate using IP.
the following string generates response 405
register => asteriskIPaddress@SIPproviderIP:5060
doing the following is not alowed by asterisk
register => @SIPproviderIP:5060
any ideas?
--
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Erick Perez
Linux User 376588
http://counter.li.org/
2005 Jan 26
0
Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 24.172.221.22
I have a PCPHONELINE SIS phone set it up to asterisk
Registered SIP '205' at 24.172.221.22 port 2770 expires 120 (Port changes
every time)
Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from
24.172.221.22(24.172.221.22 is my phones IP)
Anyone have an idea what the problem is?
Jeff
2005 May 30
1
Remote phone: Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from
One of our remote user's phone reports frequently:
Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from <IP>
What can I do ???
bye
Ronald
2014 May 09
1
Adding a SIP header to a reject 503
Is there a way to add an X-Header to hangup(34), which translates to a 503?
I tried adding it before the hangup but it never gets transmitted
2009 May 12
1
Wanting to manipulate SIP response headers
My boss has asked me if there is a way to send back a 503 response to a
request at will.
I don't see anything in the documents that would allow for manipulation
in asterisk at that low of a level.
Am I wrong?
Bruce Ferrell
2009 Jan 14
8
evaluate SIP response codes in dialplan
Hi!
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
I have an Asterisk server which forwards requests to various PSTN
gateways with SIP. If the Dial() attempt is not successful I want to
differ at least these 3 options:
- called destination is busy (486): e.g. activate auto-redial
- called destination does not exist, unassigned number (404)
- gateway is broken,
2005 Jul 22
0
Outgoing SIP causes error Got SIP response 482 "Loop Detected	 " back from.....
Hello fellow asterisk people!
I have Asterisk listening on port 5061 and SER on port 5060.
Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP.
My problems are with SIP. I can make incoming calls from SIP to asterisk
and to any of the other networks, but when I try to make an outgoing
call from Asterisk to SER I see the following in Asterisk:
-- Executing
2008 Oct 19
0
Got SIP response 603 "Declined" back from 81.15.xx.xx
Asterisk is behind firewall, I'm able to register with the provider.
Calls are coming IN OK, but when I try to call out I got:
Got SIP response 603 "Declined" back from 81.15.xx.xx
--
#Joseph
2010 Apr 10
1
Repeated: Got SIP response 489 "Bad event" back from
Hi All,
I've two asterisk servers on the same LAN, both 1.4, and I keep getting
"Got SIP response 489 "Bad event" back from 192.168.3.10"
No idea whats causing it. The only references I can find mentions NATing
issues, but these are on the same LAN so NAT shouldn't be an issue.
3.10 does authenticate into the server logging the error. The error
appears in the log
2011 Feb 17
1
Got SIP response 400 "Bad Request" back from
Hi,
I have an Asterisk 1.8.2.3 installed (public IP) with a peer (Polycom
IP601) installed behind NAT.
When the peer makes a call, it's working without any problem. But when a
call is coming back, it ends up with a Got SIP response 400 "Bad
Request" back from xx.xx.xx.xx where the xx.xx.xx.xx is the public IP of
the peer. And the call drops to the voicemail (congestion at peer
2009 Nov 23
0
Got SIP response 420 "Bad Extension" back from inphonex.com
Hello:
New to asterisk and hoping to use for http://summitcamp.org research
station.
While trying to use with Inphonex I find that incoming calls drop after
about one minute--
-- Got SIP response 420 "Bad Extension" back from 208.239.76.169
== Spawn extension (incoming-inphonex, 210, 1) exited non-zero on
'SIP/inphonex-095bf208'
Found that I can use `*CLI> sip
2010 Jul 20
0
Got SIP response 603 decline, then the call hang up
Hi to all, I have a strange behavior in my asterisk server.
I have a queue for 5 agents, the calls enter the queue an go to the agents
normally, but if I need to transfer or dial directly to an agent extension
that is already in a call, the pbx hung up the actual call (not the
transferred call).
This is what I see in the log.
Called 103
-- Agent/103 is ringing
--
2012 May 04
1
Broadvoice Got SIP response 503 Service Unavailable
Hi,
I'm running Asterisk 1.8.11.1 @office.
The Broadvoice service work fine with all 1.6 version and early 1.8
behind a NAT but about 2 months ago stop working.
No made changes in the firewall NAT rules. Right now I'm @home via my
Xlite softphone working fine without problems
Any suggestions or thoughts?
Alex Celi
This is the info
central*CLI> sip show peers
Name/username
2005 Mar 03
3
Asterisk not relaying back the SIP response messages
HI all,
I have the following setup running:
EP<--->Calling Asterisk<--->Relaying Asterisk<--->Softswitch<---> PSTN
The Endpoint EP is registered with the Calling Asterisk. Calls are
forwarded from this machine to
Relaying Asterisk which in turn forwards it to the Softswitch. In
addition, this machine also
relays back responses from the Softswitch to the Calling
2011 Mar 28
2
Dialplan help: hang up incoming call and call the number back
Hi,
I'm trying to setup Asterisk so that:
1. I call a specific number that goes to a defined extension from my
phone (an external line).
2. Asterisk notes my phone number (the CLID) and hangs up without
picking up the call.
3. Asterisk initiates a call to my phone and prompts me for a passkey.
4. Asterisk validates the passkey and lets me enter another number (say FOO).
5. Asterisk dials FOO
2004 Jun 10
4
incoming DTMF on iConnectHere?
Hi,
Anyone having problems receiving DTMF on incoming iConnectHere
lines? They disappeared for us sometime in the last 12 hours...
And, yes, we've restarted * and rebooted our * machine.
Michael Swan
Neon Software, Inc.
2004 Sep 30
1
how to hung up a call immediately if it SIP response 486 "Busy Here" received
Hi,
I noticed that it takes around 5 sec before the phone hang up
immediately if SIP response 486 "Busy Here" was received.
How to change it so that it will hangup immediately.
>From the asterisk CLI, I am seeing
ocalhost*CLI>
-- Executing Macro("SIP/6200-70bb", "oneline|SIP/6203") in new stack
-- Executing
2004 May 18
1
VoiceMailMain dumps user back into my incoming context after leaving a message
I have a dial plan that includes a company phone directory as a main menu
option. If they just sit at the main menu, after 20 seconds, they are
transferred to the operator. If the user picks an extension from the
directory, they are transferred to the proper extension. If the called
number is not available, they are transferred into VoiceMailMain. They
leave a message, and hang up. The hang
2004 Jun 18
3
WaitExten substitute
i am using the freebsd port, which seems to not yet have WaitExten(),
which i kinda want to use thusly
[ext-666]
exten => _.,1,SetVar(areacode=666)
exten => _.,2,Background(zz-in-who) ; give them list of extns
exten => _.,3,WaitExten(10) ; let them enter extn to call
include => extensions
include => applications
include => speeddials
2008 Jun 10
1
Delaying SIP disconnect after incoming call hangs up?
I'm looking for a way to delay the disconnection of a call to
a SIP extension (or pad it with silence) for a few seconds, after
an incoming call to that extension hangs up.
Rationale: I have an Asterisk PBX (current 1.4.20 codebase), with
a Leadtek BVP8051S ATA hooked to an analog phone which has a
built-in answering machine. Incoming SIP connections to the
appropriate extension are dialed