similar to: How STUN work?

I am not that good at regex and it's use in Asterisk. I am running Asterisk 1.4.13 Currently I have this in my extensions.conf for incoming calls on our house phone line: [housemenu] exten => s,1,GotoIF($["${CALLERID(num)}" = "815xxxxxxx"]?s|12); 815xxxxxxx is our home phone number, when caller id fails or is missing that is what is recorded. I want to expand this

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040729/4cfa3258/attachment.htm -------------- next part -------------- ? Hi all, I would like to study the asterisk source code(Program). I dont' know from which file i've to start reading the code. can anyone helpme. Regards Shan.

Hi, I cannot seem to accept incoming calls from FWD using IAX2. I followed the directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing calls fine using IAX via FWD. When someone calls me from FWD I get the following message: Chan_iax2.c:5251 socket_read: Reject connect attempt from 65.39.205.121 Any ideas? Thanks, S.

Annoying :( S. -----Original Message----- From: Mail Delivery System [mailto:postmaster@bembang.com] Sent: Thursday, September 02, 2004 3:06 AM To: stormp@telus.net Subject: Mail delivery failed: returning message to sender This message was created automatically by mail delivery software (HiveMail). A message that you sent could not be delivered to one of its recipients: sysad The error

Hi to all the * people out there, Please kind to me as I am both new to Asterisk and to Linux - But I am learning fast. My config is quite simple, I'm just following examples and the Wiki: I have two PC's running X-Lite phones, these work without problems between each other, and I have a GS BudgeTone-100 registered to Free World Dial UP (working no problem). I have tried to

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040728/3306e606/attachment.htm -------------- next part -------------- ? Hi, I'm dialling 1234 in the softphone or Grandstream Phone and without disconnecting the phone i want to dial "10" after dialling "1234". Is it possible to do? Regards shan

Hi, My asterisk server is behind firewall and i am trying to connect to FWD. i hv configured as mentioned in this link http://www.freeworlddialup.com/advanced/iax. i am able to register my server with FWD. But when i dial 393612, i always get 'No one is available to answer this time, try again later'. I hv portforwarded tcp 4569 and 5060 from my firewall to my asterisk server. Any idea

Hi All, I am having trouble with receiving calls from FWD via IAX. I know this isn't a FWD support forum, but I suspect the problem is my asterisk setup. The problem is that I can dial out to fwd subscribers, even myself but they can't dial me using my FWD number. I don't know much about IAX, but it would seem to me like a registration problem, but I get no errors or warnings in

I am having trouble with this. I can dial 1800 numbers fine as well as FWD service numbers but not Vonage. I can be called from ipkall and fwd and can call aixtel numbers. I use aix2 with Fwd. My extensions.conf for Vonage: ; vonage numbers ; ; +2431 exten => _2431XXXXXXXXXX,1,SetCallerID,${FWDCIDNAME} exten =>

Dear all, I have enabled stun module and configured it in asterisk , but asterisk not using stun returned public ip address for any of the sip requests going out of my network. i have done settings as below res_stun_monitor.conf settings: [general] stunaddr = stun.ideasip.com stunrefresh = 30 stun show status Hostname Port Period Retries Status ExternAddr

Hello, Recently our dell sc1425 server has been locking up with kernel freezes and required a hard reboot on each occasion. I've looked on the centos forums with limited success - each problem seems slightly different (some failure on high load, some not). Our kernel is 2.6.18-274.17.1.el5 and /var/log/messages show the following errors: Apr 3 12:41:25 sp2 kernel: INFO: task

Hi Folks, Just wondering if anyone can give me some pointers, I'm configuring Asterisk to talk to FWD's new IAX service. The asterisk server is behind an iptables NAT Firewall, with port 5036 forwarded: $IPTABLES -t nat -A PREROUTING -p udp -d $EXTERNAL_IP --dport 5036 -j DNAT --to-destination 172.16.20.200:5036 I can make outgoing calls just fine, but when I receive an inbound call

I have set up FWD via IAX service. I have tested the IAX service with 613, echo test, and 612, saytime. It all works well. However when ringing a FWD user, I got this error all the time: Connected to Asterisk CVS-v1-0-02/01/05-09:34:45 currently running on chat (pid = 8282) chat*CLI> Verbosity is at least 3 -- Executing SetCallerID("SIP/1001-a1fb", ""David

HI all! I'm looking for some infos to configure stun server support for a SIP peer. I've installed Asterisk 1.4.3, but searching for stun support in chan_sip (sip.conf) i've found nothing, only a "misterious" externip = stun... But where i have to put the ip of stun server? No infos around Google and forum! :-) Thank all, regards -- Marco Ciacci Asterisk Admin Windows

I have been trying out several stun servers with Asterisk 1.6.0.9 and 1.6.1.0 and I keep getting the following message: [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]:

is there general documentation on how asterisk behaves as a stun client (besides res_stun_monitor.conf) ? e.g.,: * can asterisk use multiple stun servers ? (im interested in availability, not data parity) * what is the relationship between gtalk.conf's stunaddr and res_stun_monitor.conf ? will duplicate queries be sent ? * Does asterisk provide some call (through AMI,

Hello I was wondering if someone knew of an application that could check that the user has a firewall and a broadband connection that will work OK with Asterisk and VoIP. The app would first perform some bandwith + jitter tests, and will then call a STUN server to check that the firewall isn't symetric. BTW, is Asterisk now STUN-capable, or is it still to map ports manually on the firewall

Having some trouble with sip behind a nat. So tried: stunaddr = numb.viagenie.ca in sip.conf. Didn't help so tried stun debug: asterisk*CLI> stun set debug on STUN Debugging Enabled STUN Packet, msg Binding Response (0101), length: 36 Found STUN Attribute Mapped Address (0001), length 8 Ignoring STUN attribute Mapped Address (0001), length 8 Found STUN Attribute Changed Address (0005),

Howdy, I have the following issue and would like to know if anyone has got around this before. IP Phones - Linksys 942 Sip server - Asterisk 1.4.13 Stun server - Vovida Ok heres the issue. We have multiple client phones on their own network behind a natted connection. We have setup the phones to be natted and also pointing to our stun server. Now when the phones make an outside call to the PSTN

Hi, All I have asterisk installed behind non-symmetric NAT, so I have NAT traversal issues. How can I use STUN or TURN to register with the other end? Or Asterisk doesn't support it? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070718/de87a89b/attachment.htm