similar to: cisco phones problems

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.