Displaying 20 results from an estimated 1000 matches similar to: "registration failure"
2004 Apr 23
6
Polycom registration
I have a PolyCom Soundpoint 500 sip phone. I'm tring to get the phone
registered on an asterisk box but am having no luck. I get the
following errors 192.168.22.196 being the phone and 22.254 being the
asterisk box..
Apr 23 11:41:33 NOTICE[1133742896]: chan_sip.c:5623 handle_request:
Registration from '"110" <sip:192.168.22.196@192.168.22.254>' failed for
2004 Apr 07
2
error 488 - Not Acceptable Here
I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone
are registered by the below information
*CLI> sip show peers
Name/username Host Mask Port Status
2002/2002 192.168.22.199 (D) 255.255.255.255 5060 Unmonitored
2001/2001 192.168.22.200 (D) 255.255.255.255 5060 Unmonitored
2000/2000 192.168.22.198 (D)
2010 Jun 15
4
can't seem to register, status unmonitored
Hi everybody,
I am trying to register my softphone(twinkle) on an asterisk server.
Everything seems to be fine.
Here is the output on show registrations in twinkle:
Tue 18:57:51
nikhil: you have the following registrations
<sip:2001 at 172.26.48.208 <sip%3A2001 at 172.26.48.208>>;expires=3013
208 is ip of the asterisk server.
on the server on doing 'sip show peers' , it
2014 Dec 30
3
status - Unmonitored, how to change it
How to change status of peers "Unmonitored" to monitored?
Home users showing "Unmonitored" some display timing.
Name/Username Host Mask Port Status
zoiper_kathy/zo 112.200.83.69 (D) 255.255.255.255 9330 Unmonitored
clinic_server (null) (D) 255.255.255.255 0 Unmonitored
voip
2011 Oct 27
5
Asterisk Executing outbound dial number twice
Hello,
I noticed Asterisk 1.8.4.1 execute number dial twice
Log
== Using SIP RTP CoS mark 5
-- Executing [912066604 at sipphones:1] Set("SIP/4773-0003e920",
"CALLERID(num)=2066604") in new stack
== Extension Changed 4773[sipphones] new state InUse for Notify User 4701
-- Executing [912066604 at sipphones:2] Dial("SIP/4773-0003e920",
2004 Dec 06
1
iax2 nativ bridge question?
hallo all,
i would like to know, as i would suspect, nativ bridiging should work also,
if only one iax party is behind an nat router and the other has a public
ip. when i connect to iax clients, which have both pubic ip's nativ
bridging is working. if one of the clients is behind an nat, the iax2
channels always get routed through the asterisk server (latest stable
version from cvs) ?? i
2005 Mar 04
2
Broadvoice + incoming call works only for ~2 minutes
Hi, all.
The asterisk setup is working fine, receiving calls via broadvoice "initially". ?
When call comes in via broadvoice number, asterisk picks it up and routes
correctly, as long as the call came in within ~2 min from the previous one.
In other words, as long as a call comes in within ~2 min since the previous one,
asterisk will answer the call. However, if the call comes in
2010 Jan 11
2
Extension Status
Hello,
I am new in Asterisk Community, i am working on Asterisk 1.6, i need to know
how can i monitor the extension status?
when i wrote sip show peers on asterisk
Extension Domain port Status
111/111 (Unspecified) D 0 Unmonitored
1300/1300 192.168.50.111 D 5060 Unmonitored
222/222
2005 May 17
4
multiple sip accounts from same sip registrar
Dear all,
I have an asterisk sip issue which I don't believe is unique.
I use a registrar (sipgate.co.uk) where I have 3 different accounts.
These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users.
By using just one of these accounts I can set asterisk up to send and receive calls no problem.
However, when I start to introduce an
2004 Jun 15
3
Grandstreams randomly go busy with Asterisk?
I've searched the lists but I didn't find anything exactly like this.
I have two Grandstream BT101 phones connected to an Asterisk.
Periodically, for reasons that I can't determine, one or the other (or
both) of the BT101s decide(s) to go on permanent busy. Dialing that
phone gives:
-- Executing Macro("SIP/24567-7856", "dialphone|SIP/27654") in new stack
2015 May 28
4
Peer is UNREACHABLE
Hi list!
I have a problem and I hope someone can help me...
I configured an Asterisk on a VM to serve more accounts and act as a proxy to
other SIP-providers.
The first account running on my phone works without any problem.
A second account, running on the phone of my wife, is always UNREACHABLE.
I can just see in the log:
[May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer
2004 Nov 21
1
make asterisk accept Register messages
Hi all,
I don't know why but my * is not accepting Register Messages.
Have you seen this kind of problem before??
I need help!
Thank in advance.
####################################################
*CLI> Nov 19 15:42:11 NOTICE[12893]: chan_sip.c:4869 register_verify: Peer
'111' is trying to register, but not configured as host=dynamic
Nov 19 15:42:11 NOTICE[12893]:
2005 Jul 15
2
[Aserisk-Users]no audio inside the net
Hi list, i've problems with my * server and the 4 phones which are
linked to it. i've 2 grandstream bt100 with the firmware upgraded to
101, a wi-fi phone (i don't know its brand) and another ip phone i
don't know its brand. with this sip.conf :
[general]
port = 5060
bindaddr = 192.168.100.229
context = default ;x changed from default to sip
localnet = 192.168.100.0/24
2007 Jul 08
2
Auto Fall Through when kicking users in MeetMe
Hi all,
My scenario is such that I have three users connected to a conference.
CLI> meetme list 1234
User #: 01 9176502096 <no name> Channel: Zap/23-1
(unmonitored)00:00:32
User #: 02 john john Channel: SIP/john-b7800468
(unmonitored) 00:00:28
User #: 03 6463875998 <no name> Channel: Zap/22-1
(unmonitored)00:00:19
3 users in that
2004 Jul 08
3
Audiocodes -> Asterisk Implementation
Anyone out there have the AudioCodes MP-108 working with Asterisk? I am able to get the channels to registers with Asterisk, but anytime I try and send a call I receive these error messages:
Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:771 __sip_ack: Stopping retransmission on '117801284512845hUxv-9991110061--17708185305@63.201.117.76' of Response 20587: Found
Jul 6 15:12:10
2007 Dec 02
4
get SIP extension status without calling it
Hi,
I am trying to get a SIP extension's status without
actually making a call.
I am using sofia-sip's "options" example utility and
the sip clients are SJphone softphones.
2004 Jul 08
2
Cisco 7960 NAT question
I've got 4 Cisco 7960's and they're behind a firewall (sonic wall). The
asterisk box is on a WAN connection on the other end of a DS3, the
phones connect fine to the Asterisk server as you can see from the
output of show sip peers below.
tp3/tp3 <firewall-ip> D N 255.255.255.255 60665
Unmonitored
tp2/tp2 <firewall-ip> D N
2005 Oct 12
8
SIP behind NAT to pub Asterisk, best solution?
What is the best solution? I dont want to have modify firewall's at all or
do port fowarding. Ideally I would like a solution that with either a
softphone or wireless hardphone one could connect via friends, family, or
hotspots without reconfiguring their devices.
What are people using? STUN? SER?
Thanks in advance!
-blake
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2008 May 02
2
sip show peers
When doing a "sip show peers" I might see something like:
Name/username Host Dyn Nat ACL Port
Status
devcentos5x64_to_mmfirepa 192.168.1.177 5060
Unmonitored
devcentos5x64_to_bt610tMM 192.168.1.159 5060
Unmonitored
devcentos5x64_to_am2mm/de 192.168.1.178 5060
2012 Jul 12
1
Asterisk with OpenBTS and mobile phone
Hello mailinglist,
I want to connect Asterisk with OpenBTS and make a call with a mobile
phone.
I use:
Ubuntu 11.10 + Kernel 3.0.22
GnuRadio 3.3.0
Asterisk 1.8.13
OpenBTS 2.8
Nokia Mobile Phone
OpenBTS works and I can send sms from the OpenBTS server to the
mobile phone. What I also need is a call between Asterisk and OpenBTS.
I have also two soft phones which works with Asterisk. And also