similar to: Snom 200 errors?

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]:

(hardware in my computer: linux, asterisk, x100p, grandstream budge tone-100 ) Hi, When i run #asterisk ?v It show me a messages but when i try to incomming the call it show me that. Apr 11 07:59:01 NOTICE[81926]: chan_sip.c:3140 sip_reg_timeout: Registration for 'me@192.168.0.6' timed out, trying again Apr 11 07:59:01 NOTICE[81926]: chan_sip.c:5568 handle_request: Registration

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

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

I'm wondering if anyone else has gotten something similer to this? I had FWD working fine on the asterisk box, then all of a sudden it just stopped working. I get the following errors (just keeps looping) *CLI> DEBUG[1125329600]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '6dc8436c7c568eea75fffdc75478ed54@142.55.31.179' of Request 102: Found

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

What does this mean? Asterisk 1.2.13 talking to 1.4.0. (response from 1.4.0.) Yuan Liu

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

What does that mean? Where can I get more info about it? Mar 18 07:19:51 WARNING[9563]: chan_sip.c:7549 handle_response: Got 200 OK on REGISTER that isn't a register

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

I am having a problem when a SIP registration fails. I get the following messages in the log: Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 2874 (sip_reg_timeout): Registration for '<user>@fwd.pulver.com@65.39.205.114' timed out, trying again Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 5119 (handle_request): Registration from

When run asterisk ?vvvgc IT show me this error Asterisk Ready. *CLI> Apr 11 08:59:27 NOTICE[81926]: chan_sip.c:3140 sip_reg_timeout: Registration for 'phone@192.168.0.6' timed out, trying again Apr 11 08:59:27 NOTICE[81926]: chan_sip.c:5568 handle_request: Registration from '<sip:phone@192.168.0.6>' failed for '192.168.0.6' Apr 11 08:59:27 NOTICE[81926]:

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

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

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

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",

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

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031214/949c1368/attachment.htm -------------- next part -------------- Hello I have configured IP 7905G and SNOM 200 for Asterisk. Now problem is that I can call from IP 7905G to SNOM 200 but not the other way round. Instead I get "FORBIDDEN" Message on SNOM 200 LCD when ever I try

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031215/9327b656/attachment.htm -------------- next part -------------- Hello I have configured IP 7905G and SNOM 200 for Asterisk. Now problem is that I can call from IP 7905G to SNOM 200 but not the other way round. Instead I get "FORBIDDEN" Message on SNOM 200 LCD when ever I try