similar to: Repeated Notice:

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'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

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

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,

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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

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

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 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

I'm having trouble setting up tls/srtp secure communications on my Asterisk server- I'm still rather new at working with Asterisk. I have enabled tls and encryption and I have csipsimple with tls build on the phone. I'm currently only testing one phone with this capability so far, and the rest still work in the current state. My logging looks like this with verbose turned up:

I have 1 IP phone (Swissvoice IP10S) and 1 POTS phone. When I dial the number for the IP phone off the POTS phone, the IP phone rings. But when I pick up the handset on the IP phone, I get a busy signal and this message on *: Aug 23 09:38:57 NOTICE[1142106560]: chan_mgcp.c:2243 handle_response: Terminating on result 502 from svip10@00059002042b-1 Here is the entire session. svip10 is the 1 and

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

Hello, Hope everyone is enjoying their holiday! We setup two asterisk servers (From CVS on Wednesday) and set up IAX between the two. Right now they both reside on a switch with a static 192.168.0.x IP address. The first Server is .5 and the second is .30. Our dialplan seems to be working, however on the console we get a flurry of NOTICE and WARNING messages. NOTICE[1116941120]: File

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

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

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

I have been playing with asterisk for a couple of weeks now and I have been very happy with its performance. However, I have run into a problem with how I want to deploy this solution. I have a mix of softphones (SJ and Xlite), ATA's, and a couple of IP phones (Snom 190). The asterisk box is on the public network. For my primary users they will reside behind a watchguard 4500 firewall.

Hello, I have a local Asterisk server that keep loosing its registration to main Asterisk server. The local asterisk server is on the local subnet, it acts as a client with extension 808. Local server Sip.conf register => 808:password at as2.xxxxx.com registertimeout=20 registerattempts=10 Main Asterisk Server sip.conf [808] type=friend context=sip-phones call-limit=99