search for: cellroute

...t - main/logger.c:ast_log >> # Address format - ast_sockaddr_stringify >> # >> # First regex: channels/chan_sip.c >> # >> # main/logger.c:ast_log_vsyslog - "in {functionname}:" only occurs in s >> > > > -- > Technical Support > http://www.cellroute.net > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hell...

...ceived after 0.844 ms ** > SIP/2.0 404 Not Found > final received > thufir at doge:~$ > > > However, I'm sure you're right that it's the dialplan; I'm looking into > it. > > > thanks, > > Thufir > > -- Technical Support http://www.cellroute.net

...ge, "hi", logged anywhere? I don't think so. But you should also see the SIP messages on the console (sip set debug on) without having to look at the log file. Maybe something in your logger.conf is messed up. > > > > -Thufir > > -- Technical Support http://www.cellroute.net

...gt; thwart any >> attempt at interoperability is truly shocking. >> >> For connecting two Asterisk installations to each other over the >> Internet, IAX >> is better than SIP -- that's what it was designed for. >> > > -- Technical Support http://www.cellroute.net

hello list, i have a question i don't know if there is any possibility to stop asterisk using a call for exp: when i call a number 0522xxxxxx i want to excute a script or any idea to stop asterisk automatically i use asterisk 1.4.43 NB: with mysql using a database i can insert into table using php without issue. but now with SSH how can i do thanks and regards. -------------- next part

Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f Thank you! -- Allan Porras http://allanPorras.com <http://www.AllanPorras.com> Google Plus: http://goo.gl/BRkbX Twitter: @alpocr <http://twitter/alpocr> --------------

Hi all, I'm having issues with overwrite caller id, when I call someone my caller id should be "mycompanyinc" but instead my id shows up as my extension number 101. this is what i have in sip.conf [101] type=friend context=sipphones call-limit=99 callerid="iuser 101" disallow=all allow=ulaw allow=alaw username=101 secret=Passwd dtmfmode=rfc2833 host=dynamic mailbox=101 at

Hello, I have an asterisk box with a peer configured with nat=force_rport,comedia, but asterisk keeps sending the audio to the private IP address and ignoring the client peer nat settings. If I check the "sip show peer extension", I see both symmetric RTP and Force Rport are set to yes, but asterisk seems ignoring them. Force rport : Yes Symmetric RTP: Yes Asterisk is behind a

.../2.0 404 Not Found > final received > thufir at doge:~$ A "sip set debug on" will give you more info on why you are getting the 404. It probably has to do something with your context/dialplan. > > > thanks, > > Thufir > > -- Technical Support http://www.cellroute.net

Hi guys Thanks for the pointers - I'll look into the possible compromise scenario though I've got no idea how I'll counter it -if- I manage to detect it...! I've disabled prelinking (thanks Tony!) and I'll see if that helps. Interesting thing I've now discovered (had this failure again at the head office this morning) is the "growth" in the file's size is

Hi all, I have an SPA112 that in sitting behind a Ubee cable modem. The internet link is solid, but the device becomes unreachable within a day or so of being rebooted. Then the customer goes to reboot the device, they report that all 4 lights are lit. The ISP reports that the device does respond to ping, so it's not completely dead. I've had the same symptoms with SPA303's

On Fri, 20 Feb 2015 08:46:13 -0500, Andres wrote: > A "sip set debug on" will give you more info on why you are getting the > 404. It probably has to do something with your context/dialplan. on tleilax: tleilax*CLI> tleilax*CLI> sip set debug on SIP Debugging enabled tleilax*CLI> on doge: thufir at doge:~$ thufir at doge:~$ sudo sipsak -vv -s sip:devries at

Hello I'm using the Fail2ban. I configuration below. I want to try to prevent the continuous password. Fail2ban password that does not prevent this form. (Asterisk 1.8 / Elastix interface) What could be the problem ? Asterisk log; "Registration from '<sip:3060 at sip.x.eu;transport=UDP>' failed for 'x.x.x.x:32956' - Wrong password" Fail2ban asterisk

...ts me an >> email saying there was an update. It's a fairly simple and >> straightforward process and does the job. I use this script for all >> PBX?s that are behind a NAT. I hope this helps. >> Regards; >> John > > > -- Technical Support http://www.cellroute.net -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150401/91a58518/attachment.html>

On Fri, 20 Feb 2015 15:07:35 -0500, Andres wrote: > This is showing nothing so I don't think your test message even made it > here. I think it looped in the 'doge' server. I was wondering the same thing :) in tleilax, I looked in /var/log/asterisk/messages and see: [Feb 20 15:13:19] VERBOSE[3661] chan_sip.c: [Feb 20 15:13:19] <--- SIP read from UDP:192.168.1.3:38154

I have an asterisk 1.8.x box that intermittently returns a 401. Calls come through the same peer all the time, from the same carrier. However intermittently the asterisk box returns a 401. Below is the output of a failed call (1st) and a successful call (2nd). I can't see any difference until we get to these lines. Bad call: --- (17 headers 14 lines) --- Sending to carrierIP:5060 (no NAT)

Hi list! Another day, another problem... I'm checking with Nagios my Asterisk at home, and since yesterday I noticed that, after my IP changes (Deutsche Telekom drop the DSL-line every 24 hours, so that I have a new IP every day), the peer of an VoIP-provider I use is UNREACHABLE. Yesterday I though it was a problem on the line, but today is the same, so I think it is something other...

I am not proxying the media, but never the less I am forced to store the source media IP in my CDR, for regulatory reasons. Asterisk gets that information when the reinvite comes, but how do I store it? If I don't figure this out my next email will be from Federal Prison. Kindly help me stay away from those guys. Eventually we all need to save that information or we shall not be able to stay

I?ve gotten to the bottom of this; Seems that the pjsip.endpoint_custom.conf isn?t getting included properly, or my syntax is wrong. If I put force_rport=no into pjsip.endpoint.conf and reload only Asterisk, everything works perfectly. Unfortunately, I?m using FreePBX, so it owns this file and my changes won?t persist a FreePBX reload. Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F

Hi All. Someone is attacking on my SIP server. There are lot of requests coming in and I am not able to stop it because I am unable to detect the IP address. I used wireshark to capture the packets. Although I am using very strong password for my SIP users but still is there any way to drop these packets and stop this attack. I tried dropping packet after matching some string (most of the