Displaying 20 results from an estimated 3000 matches similar to: "cisco phones problems"
2004 Jun 22
5
CISCO 7960 Goes missing
I've got a number (10) Cisco 7960's connected to my network. All the phones
work perfectly except one.
The asterisk console keeps throwing up:
Jun 22 15:39:15 NOTICE[-1147470928]: chan_sip.c:5887 sip_poke_noanswer: Peer
'4001' is now UNREACHABLE!
Jun 22 15:39:27 NOTICE[-1147470928]: chan_sip.c:4925 handle_response: Peer
'4001' is now REACHABLE!
Jun 22 15:42:08
2004 Sep 20
5
iax2_read: I should never be called
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2004 Apr 20
1
Repeated Notice:
I see repeated over and over the following messages:
NOTICE[1142106560]: chan_sip.c:4988 handle_response: Peer '1001' is now REACHABLE
then 5 minutes later:
NOTICE[1142106560]: chan_sip.c:5958 sip_poke_noanswer: Peer '1001' is now UNREACHABLE
both messages repeated over and over
Any clue what I can do to fix this?
Is there any where I can look up these Notices to find
2004 Sep 03
1
SIP / Keep alive...
Hello list,
Is there some parameter on sip.conf to always let the client reachable ?
I'm trying to avoid .... this situation :
Sep 3 09:49:29 NOTICE[135442432]: chan_sip.c:7653 sip_poke_noanswer:
Peer '1264' is now UNREACHABLE!
Sep 3 09:49:39 NOTICE[135442432]: chan_sip.c:6408 handle_response: Peer
'1264' is now REACHABLE!
Regards,
-Jefferson Carvalho
2004 May 23
1
IAX2 REACHABLE/UNREACHABLE
All,
I have an issue with IAX that I can't comprehend. Approximately every eight
minutes my servers go unreachable. They stay unreachable for exactly 10ms.
I have two servers running IAX and it happens on both servers
simultaneously. I have searched the archives and see similar issues, but
not the exact same one. I am on the current CVS stable version of *.
Also, during IAX calls,
2005 Jul 27
5
does not implement 'PUBLISH'
Not sure what this is.
When I call my own ext the call will ring for 10 sec and goto the
voicemail. However the phone will keep ringing and I see this on the
asterisk CLI
Jul 27 11:17:48 WARNING[3563]: chan_sip.c:8666 handle_response: Host
'192.168.0.200' does not implement 'PUBLISH'
Have no idea what this is talking about
192.168.0.200 is a cisco 7960G
2008 May 01
1
Remote host can't match request NOTIFY???
Hi all,
I'm seeing a lot of these messages:
[Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response:
Remote host can't match request NOTIFY to call
'2e4a02807750b7024d5ff09c668fa0f7 at 10.0.0.2'. Giving up.
[Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response:
Remote host can't match request NOTIFY to call
'0755ad8f40b9d09d491b635e70bb8905 at
2007 Mar 22
2
Asterisk 1.4.2
Hi all,
I upgrade my asterisk from 1.2.11 to 1.4.2 changing my entire dial plan
but I have the following errors and I'm not able to call anymore. Do you
know what can I have to do?
My Asterisk is connected to a patton with a SIP trunk.
[Mar 22 10:19:03] WARNING[16462]: chan_sip.c:12311 handle_response:
Remote host can't match request BYE to call
2004 Jul 31
3
MGCP & Cisco ATA 186 Help
Does anybody has the expirience configuring Asterisk with Cisco ATA 186
MGCP firmware ?
I have Cisco software v3.1.1 atamgcp (Build 040629A)
Asterisk 1.0-RC1
On ATA i only put domain test.
mgcp.conf looks like this
[test]
host = 192.168.195.55
context = default
line => aaln/2
line => aaln/1
Asterisk CLI shows this:
Jul 31 16:05:40 WARNING[135449600]: chan_mgcp.c:485
2005 Sep 23
2
asterisk invitation problem
when i send calls from an asterisk box to a voip
provider the call fails and give me these messages:
*CLI> Sep 23 21:32:32 WARNING[14595]: chan_sip.c:6890
handle_response: Forbidden - wrong password on
authentication for INVITE to '"asterisk"
<sip:asterisk@195.112.214.99:5070>;tag=as19e688a1'
-- SIP/call-0f60 is circuit-busy
== Everyone is busy/congested at this
2004 Jan 06
7
911 and lawsuits
Just curious if any of the Asterisk installers are doing anything special
to protect themselves from a possible lawsuit caused by 911 failure
during a Asterisk/computer crash?
I realize that any traditional PBX or even a phone line can fail but,
anything running on a computer is probably going to be less reliable
than most PBXs.
Anybody requiring customers to acknowledge and sign any kind of
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
2009 Apr 07
1
i have a probleme and my asterisk and ovh
hello every body
my connexion on ovh to pass in UNREACHABLE and not reidentified were not
reboot the server.
[Apr 7 20:17:21] NOTICE[19947]: chan_sip.c:15605
handle_response_peerpoke: Peer 'ovh' is now Lagged. (2067ms / 2000ms)
[Apr 7 20:17:35] NOTICE[19947]: chan_sip.c:19829 sip_poke_noanswer:
Peer 'ovh' is now UNREACHABLE! Last qualify: 2067
but my probleme is the adress
2004 Aug 16
3
Formatting in sip.conf...can you have 2 @ signs for register?
Hi All,
I am trying to setup another sip trunk in addition to what I am already using. The sip provider we are using right now gives you your username as your email address. So IE. If my email is james@james.com.... that is my username . Now... When I put this in the sip.conf file I have found that Asterisk is not able to parse it correctly and instantly goes to the email server to authenticate
2008 Sep 22
1
I can't call my remote users?
Good day to all--
First off let me say that I have been very pleased with the mailing
list. I have learned a ton of stuff just reading other peoples
questions and comments. I really enjoyed the VOIP Conference call on
Friday morning. Still working on figuring out the best approach to
custom voicemail emails (the reason I joined this group); however, we
have more pressing issues. I
2005 Aug 21
2
Broadvoice Issue
I did a quick google search of the lists site and couldn't find a
definitive answer, so if it's there, I apologize for asking again.
Starting about noon yesterday, I am no longer able to send/receive calls
via Broadvoice. When calling in, I get a fast busy, and when calling out
I get the following error:
-- Executing Dial("SIP/112-572a",
2006 Jan 25
2
Voipbuster/voipstunt -- what a crap service
Hi, all
I am reallty pissed with their service. I wonder if this is common problem.
Firstly, all of my calls are terminated after 30s. And termination happens
in a strange way. My local asterisk server does not see the disconnection,
but remote party is disconnected. Basically, I am still on the phone, while
remote party was disconnected. When I hang up, I get something like that:
Apr 20
2008 Jun 20
15
before_save model callback rspec testing
hi all,
i''m learning rspec and i can''t figure out how to test if a callback is
executed in a model.
my model code is:
class User < ActiveRecord::Base
before_save :encrypt_password
...
def encrypt(password)
self.class.encrypt(password, salt)
end
thanks a lot,
cs.
--
Posted via http://www.ruby-forum.com/.
2008 Feb 13
2
MWI problem with Siemens Gigaset S675 IP
Hi list,
Before purchasing a number of Siemens DECT SIP phones, the Gigaset
S675 IP, I read that the problems with MWI had been fixed with the
latest firmware version (see
http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP). Now I'm
not so sure that's the case.
After setting up a network mailbox for one of these phones, as well as
an Asterisk voicemail account (ext.
2008 Nov 28
1
Windows Mobile 6 SIP client: Remote host can't match request NOTIFY to call
I'm trying to get my Windows Mobile 6 phone working as an asterisk client.
Overall things are working well. However, I regularly get the following
message:
[Nov 27 21:57:28] WARNING[4507]: chan_sip.c:12892 handle_response: Remote
host can't match request NOTIFY to call
'67aa8e223479055656161bf17ebb77d5 at 172.31.253.4'. Giving up.