similar to: Firewalling and Asterisk

Hello Folks; Again, excuse my cluelessness. I have an Asterisk server in the US - and I want to connect it to one in Europe. Here is my scenario: 1. call a phone number, my Asterisk box in the US answers 2. perhaps a 'please wait' voice message 3. it dials an extension on the other Asterisk box in Europe. I am not looking for someone to do this for me, I am just not really

Hey; Anyone see this before: [Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have <6839>, digest has <3169> G

Hey Everyone; I inherited an Asterisk box where the dialplan is a real mess. ( I would actually be embarrassed to post some of the stuff!) So, here is what I need to do - and again, I am looking for fishing nets and places to cast them - if I don't figure it out, I will never freakin' learn! I have several users configured (101, 102, 105, 155, 211, etc). They are all in different

Over the last two weeks, we have had at least two "incidents" where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further).

Let me explain: When I dial into Asterisk ( I have a SIP trunk - which I need to make sure is not faulty), I only get one-way voice communication. The calling party, from the SIP trunk hears nothing - the extension rings on the Asterisk server (you can see it in the CLI and hear it at the computer), and the softphone rings However, when you answer the SIP softphone , you can only hear the

Hello Everyone; I am looking for a solution to record calls that come into our Asterisk server. I am hoping for something that is easy to use - however, if I have to modify it to make it easier to use, I do not mind. Does anyone know of any opensource or otherwise solutions out there that I can try out? Thanks much. Glen

Hello Folks; I appreciate all of the help so far - thanks. Another question: I am using MixMonitor() to record calls and I would like to include the called number/extension in the filename: In my dialplan, I am able to save the file with the caller id in the filename. However, what I am a little unsure about is the incoming number/called number/extension - passing that information on to part

Hello Folks; Perhaps I am chasing my tail here. Before I go any further, is this compatible/supported in Asterisk 1.6x? If so, I would be willing to post any manager.conf or http.conf snippets needed. When I attempt to open the Asterisk Web GUI, I get a 'page not found'. I am sure this is something really minor - something silly that I missed. Any words of wisdom? Glen

Hi everyone, I want to issue the command: iptables -F and then rebuild everything from the beginning with a very limited scope and then without locking myself block all other traffic. Can you suggest what I should put in the shell that would get me this: Allow traffic from subnet 172.16.0.0/24 (my VPN tunnels) - All traffic including those of Asterisk and HTTP - I trust this network Allow

Dan et al; This looks like a perfect solution. However, I have one issue. If I initiate the macro manually (put it in the proper context/dialplan) it works. I see the *.wav file being created and growing in the /var/spool/asterisk/monitor directory. If I try to implement it adding the MixMonApp => *1,self/both,Macro,mixmon to the [applicationmap] in features.conf, I cannot get it to

With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF} is [un]set in an odd way. for example consider: 999,1,Swift(some long message that you dont want to wait for|5000|5) 999,n,NoOp(DTMF: ${SWIFT_DTMF}) if while I am listening to the playback, i interrupt and dial: - "12345", SWIFT_DTMF is set to

I upgraded from a perfectly working 1.6.2 asterisk installation (including fax via app_fax_digium) to 1.8.0 this evening. All my custom modules (including swift <thanks darren!>) are working fine except for fax. When a caller connects, asterisk switches to the fax context and hangs up the call. i've captured with: core set verbose 10 core set debug 10 fax set debug on sip

running Asterisk 1.8.9.0-rc2, what are the ways to interface with ConfBridge ? I see the CLI command 'confbridge' documented for asterisk 10, but i dont see how to interface with confbridge on 1.8 What I'm trying to do is keep track of conferences that are used. I tried something like the below, but not only does Confbridge not return, but i'd need something that erases the

Any ideas on how asterisk could accept an email (such as an email to SMS or "number at mybox.org" sort of thing) and make a phone call to a specific number and make an announcement? I imagine the first part is the big question. joe a.

I recently noticed that asterisk is not logging unknown sip connections. I'm not sure if I've broken something or if asterisk itself has been broken. the last entry I have is: /var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c: Registration from '<sip:22942 at 10.0.0.3>' failed for '10.0.0.228:5060' - No matching peer found my logger.conf

I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10. The system itself is happy and phone calls (between two parties) seem fine. Unfortunately, when a caller listens to a Playback recording, there seems to be moments of stutter - perhaps 1 second of stutter for every 10 seconds of Playback. The stutter is not consistent at the same point of the playback file. To

I have asterisk 1.6.2.11 talking to a cisco 1760 running ipvoicek9-mz.124-25b. whenever a call goes through the 1760's FXO or FXS (in or out) there is a 915 second maximum call time due to asterisk hanging up the call because of a "critical packet" being missed. I read doc/sip-retransmit.txt and I don't see anything there that is helpful to my situation - the asterisk box is

After upgrading to Asterisk 1.8.0, I am finding that my outbound SIP calls are being destroyed after 1 minute and 20 seconds (80 seconds). Asterisk is sending a BYE message - I have no idea why. http://jeremy.kister.net/tmp/20101115/sip.txt for a sip debug. can anyone suggest how i can further deal with this? -- Jeremy Kister http://jeremy.kister.net./

I recently configured a SIP peer which i must specify my fromuser as my ten digit "DID". I send calls to this peer, but whenever Asterisk sends an options message, the fromuser is "asterisk". Is this a bug? Or is there some other config I must make ? register = 2155551941:123456 at 10.0.138.226/2155551941~600 [peer](!) type=peer context=inbound qualify=yes

i'm trying to review issues that i'm monitoring and/or have reported at http://issues.asterisk.org when I click on 'My View' or 'View Issues' I get an error: APPLICATION ERROR #401 Database query failed. Error received from database was #1142: DELETE command denied to user 'mantisreadonly'@'localhost' for table 'mantis_tokens_table' for the