similar to: CISCO 7960 Goes missing

hi folks. we recently deployed 10 Cisco 7960G w/ SIP firmware 7.3 on our network and we start having problems of dropping calls (actually the calls wasn't dropped it just the sound was muted for about 5-10 seconds, but most users will think the call dropped and hangup/redial). i've check the console output. there was a lot of messages like the following: Sep 28 15:00:49 NOTICE[8182]:

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

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

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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,

Does anyone in Asterisk land know how to turn on the message light on the back of the earpiece of a cisco 7960 when a message is waiting? Thanks! Paul Paul Mahler mail:pmahler@signate.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040130/0efacc79/attachment.htm

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

Hi, I thought I remember seeing somewhere on the Asterisk website a card that had 16 ports fxo or fxs, that was user selectable with straps on the card. Am I going crazy, I can't seem to find it now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040930/86c8ae69/attachment.htm

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

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

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

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

Maybe someone could point in the right direction. I have a small facility that's running around 40 Polycom 301/501 phones, Asterisk 1.4.18 running under Mandriva 2007.1. The phones were assigned a DHCP address in the 10.10.10.x range. Today, the DHCP server failed and to get them back online, I loaded the dhcp-server onto another system (Also running Mandriva) and copied the dhcpd.conf

Hello, I am logging directly into file and also to syslog. Here is snippet from my /etc/asterisk/logger.conf: messages => notice,warning,error,verbose syslog.local0 => notice,warning,error,verbose But the logs look different: VERBOSE[7609][C-00000013] pbx.c: NOTICE[3042] chan_sip.c: Peer '1111' is now UNREACHABLE! vs. VERBOSE[7609][C-00000013]: pbx.c:2925 in

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

Is it possible to get a few 7960's and asterisk to allow all of the 7960 phones to use one extentsion and can only be used by one person at a time, have it indicate on the other 7960's when one of the others has the line engaged. Basicly so like I can setup a rule when an incoming call comes from IAX to divert to this extension, it will ring the extension (thus all phones), and allow me to

Might be a stupid question, but is there a default gateway set on the 7960? -----Original Message----- From: Paul Mahler [mailto:pmahler@signate.com] Sent: Thursday, December 18, 2003 7:04 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 - can't traverse NAT? I have a 7960 running behind a firewall running NAT. From a telnet session to the 7960, I can't ping

I am just starting to deploy asterisk in our office to use as our primary phone system - we plan to use a Voicetronix OpenLine4 card as our PSTN gateway - but one thing at a time... haven't got that far yet. Currently, i'm trying simple IP to IP calls within the office using our Cisco 7960's phones running SIP. When I make a call between these two phones, the conversation is of a

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

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