Displaying 20 results from an estimated 1000 matches similar to: "SIP registration fails"
2009 Jul 01
2
Registrations problems to SIP-provider.
Hello List,
I'm having problems with registrating my Asterisk-server to the
SIP-provider. Yesterday all worked fine, this evening I cannot call out.
What can be wrong ?
This is my registration in sip.conf :
register => 092779077:XXXX at 85.119.188.3
This the output of SIP show peers :
asterisk*CLI> sip show peers
Name/username Host Dyn Nat ACL Port
Status
2009 Jun 27
1
2 problems I can't solve without any help
Problem 1 :
Incoming conversations from the SIP-provider come into the
[default]-context and to the 's'-extension.
I am unable to change this, even if I have :
sip.conf
[general]
;context=default ; Default context for incoming calls
register => 092779077:XXXX at 85.119.188.3
; incoming
[092779077]
type=user
host=85.119.188.3
context=from3starsnet
So I define no
2009 Oct 08
2
How to keep difference between 2 SIP-accounts/trunks from same server ??
Hey list,
I have a problem when I host 2 SIP-accounts on the same Asterisk-server.
Asterisk picks out the SIP-account on alphabetic order A --> Z.
In my sip.conf :
register => user1:passwd1 at server/user1
register => user2:passwd2 at server/user2
[YOCAN-3starsnet]
type=peer
host=server
username=user1
secret=passwd1
fromuser=user1
accountcode=user1_in
[ITCENTER-3starsnet]
type=peer
2009 Jun 23
1
SIP 482 Loop detected
-- Executing [0473775006 at intern:1] NoOp("SIP/twinkle-088e6ea8",
"conversation to GSM") in new stack
-- Executing [0473775006 at intern:2] Dial("SIP/twinkle-088e6ea8",
"SIP/3starsnet/0473775006") in new stack
-- Called 3starsnet/0473775006
-- Got SIP response 482 "Loop Detected" back from 85.119.188.3
-- Now forwarding
2009 Jul 06
2
SIP registry fails during night
Every morning I check my SIP registry to the SIP-provider. And I must
conclude that during the night somewhere registry has failed.
asterisk*CLI> sip show registry
Host Username Refresh State
Reg.Time
85.119.188.3:5060 092779077 105 Failed Sun,
05 Jul 2009 23:11:40
asterisk*CLI> sip reload
[Jul 6 10:30:43]
2005 Feb 20
1
Adtran Total Access MGCP Config?
I've never set up an mgcp device before. I have an Adtran IAD with the
MGCP firmware on it. I have it configured in mgcp.conf like this:
[general]
port = 2427
bindaddr = 0.0.0.0
[adtran]
host = 192.168.2.2
context = default
canreinvite = no
line => aaln/1
line => aaln/2
The device is configured like this:
MGCP Configuration | Standard MGCP 0.1 / NCS 1.0
MGCP Endpoint
2004 Apr 23
3
Problem With zaphfc
I've this error
How i can find the problem?
Apr 23 12:24:43 WARNING[131081]: PRI: received TEI check request for TEI = 89
Apr 23 12:24:47 WARNING[131081]: PRI: received TEI check request for TEI = 89
Apr 23 12:24:48 WARNING[131081]: PRI: !! Got a UA, but i'm in state 1
Apr 23 12:24:53 WARNING[131081]: PRI: received TEI check request for TEI = 89
Apr 23 12:25:02 WARNING[131081]: PRI:
2011 Jan 18
1
Ongoing problem with 1.8
I have tried installing many of the beta versions and most of the
release versions of 1.8. I have 3 SIP phones which we use for all our
calls. After installing 1.8 the first thing I try is calling out port
one of my Digium TDM04 back into port 2. I can see that the call
comes in and tries to call all three SIP phones but the phones never
ring. Eventually the call goes to voice mail and these
2003 Aug 22
0
"Frame rejections" on E1 trucks
Hi-
I've posted this on the bugs list, but I'd also like to see if others have
had similar problems when connecting via E1 trunks (E400P).
I'm getting numerous errors like the following during inbound calls to my E1
channels. These occur when the system is under medium load:
WARNING[196621]: File chan_zap.c, Line 5404 (zt_pri_error): PRI: !! Got
reject for frame 78, retransmitting
2007 Apr 26
0
Static in Audio PRI, Got reject for frame 39, retransmitting frame 39 now, updating n_r!
Anyone know what would cause this error?
!! Got reject for frame 39, retransmitting frame 39 now, updating n_r!
!! Got reject for frame 39, retransmitting frame 40 now, updating n_r!
I assume this would cause audio issues as well.
Thanks,
Steve
> Message type: CALL PROCEEDING (2)
> [18 03 a9 83 89]
> Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
2005 Jan 07
0
Sip Phone Won't Login...
Hey Peoples,
I just got my paws on a KE1020A Phone and all it is doing when I plug it in is:
1201
Wait Login...
Sip.conf
[1201]
type=friend
username=1201
secret=<password>
host=216.254.10.183
mailbox=1201
context=intern
canreinvite=yes
dtmfmode=rfc2833
nat=1
register => 1201:<password>@216.254.10.183/1201
One side note, The KE1020A does not have NAT capabilities, but I am
2009 Apr 30
0
Asterisk and Shoretel integration
Hello everybody.
I have a problem with an integration between an Asterisk (1.4.24.1) on
FreeBSD 7.0 and a Shoretel 7.5 server.
To make a very long story short, when someone behind asterisk call an
extension behing shoretel everything work as expected. When someone
behing the shoretel server call someone behind asterisk the first 10
seconds of the call seems ok but then the line is dropped
2008 Oct 09
2
retransmitting NAT
Hi,
What does retransmitting NAT means? I have a client that uses SPA 942,
and his phone sometimes cannot be called. i did a sip sebug and i keep
on seeing retransmitting NAT.
on the realtime it shows that it is registered, so when i try to call it
, asterisk thinks it is still online so it tries to reach it instead of
saying it's unavailable,
[Oct 9 11:10:33] -- Called 103100
it
2003 May 25
1
iconnecthere problem 481 "Call Leg/Transaction Does Not Exist"
Hi All,
I am trying to use iconnecthere to make outbound calls. I am behind a
linksys router. I keep getting this error
481 "Call Leg/Transaction Does Not Exist". Does anyone have any prior
experience with this problem. Any leads will be much appreciated. Attached
are the conf files and logs
#SIP.CONF
; SIP Configuration for Asterisk
[general]
port = 5060 ; Port
2004 Jul 20
1
Random Dropped Called
I've got a 4 port T1 card in my Asterisk box with a PRI from Qwest as
my PSTN interface. I'm experiencing random dropped calls on the
various SIP devices I have tested. Network connectivity to the SIP
devices looks ok, and I have tried a variety of the devices including
all of the following.
Grandstream 286
Grandstresm 486
Sipura SPA 1000
Mediatrix 2102
Some example lines from my logs
2003 Sep 27
0
More Sip/Grandstream issues
I just checkout the cvs code for asterisk......
when I use my grandstream phone (that worked on the old code that was
about 2 months old) I do not hear anything at all...
I get this error:
Sep 27 23:20:27 WARNING[1142127920]: File chan_sip.c, Line 444
(retrans_pkt): Maximum retries exceeded on call
0765c89e-9d67-3c0a-b9b9-2e7f3cd1d9ef@192.168.50.248 for seqno 58430
(Response)
here is my
2009 Oct 10
1
Grandstream GXP 2010 : multiple accounts not working
On my Grandstream GXP 2010 I have the possibility for 6 channels and
thus 6 different accounts...
Line 1 I define an account that registers directly to an online
Asterisk-server, somewhere in a datacentre.
Line 2 I define an account that registers to the local Asterisk-server
(NSLU2 unslung)
When I activate both accounts, only the first account (to the
Asterisk-server on the internet) registers.
2009 Oct 03
0
ERROR[1499]: rtp.c:2482 ast_rtcp_write_sr: RTCP SR transmission error
Hello list !
SETUP :
Grandstream --sip--> Local Asterisk (NSLU) --iax--> Hosted Asterisk
(VirtualDedicatedServer) --sip--> SIPprovider --> my CellPhone
PROBLEM :
I've noticed that when I put down the horn of my Grandstream it still
takes a while for my GSM/CellPhone to stop ringing.
INFORMATION :
This is the output on the CLI of the local Asterisk-server :
[Oct 3 17:40:33]
2010 Dec 06
0
Fw: Sip Hangup after critical packet SIP DEBUG attached
?
HI
?
I have asterisk 1.6.13 running. My ATA are connected via vpn (openvpn)
?
Out going calls from asterisk to the ata works fine
Incoming calls from the ata to asterisk cuts off with the error msg
?
Maximum retries exceeded on transmission 70854efe-4157e3a8 at 10.168.7.103 for
seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
[Dec? 6 13:52:43] WARNING[3921]: chan_sip.c:3858
2006 Apr 02
2
Cisco 7960 nat problems.
I have a asterisk server running on site listening on a public ip. Tonight I attempted to connect a Cisco 7960 phone from my home location via sip but failed. My home network is simple, Cox cable connection hooked to a linksys wrt router. The firewall on the linksys router is disabled and I even setup dmz to the phones ip as a last resort. I removed the linksys router and plugged the phone