[asterisk-users] RTP port ranges

2013-09-13 Thread Jonas Kellens
Hello, I have defined that I want to receive audio (RTP) on port 11500 till 11954 (rtp.conf). The same range I have defined in my firewall. I now see that an IP-address gets blocked by my firewall because there are packets coming onto port 11955. How come the client sends audio on port

Re: [asterisk-users] RTP port ranges

2013-09-13 Thread Andrew Colin
Normally you should open ports 1-2 udp On 9/13/2013 11:37 AM, Jonas Kellens wrote: I now see that an IP-address gets blocked by my firewall because there are packets coming onto port 11955. -- _ -- Bandwidth and

Re: [asterisk-users] RTP port ranges

2013-09-13 Thread Jonas Kellens
On 09/13/2013 11:41 AM, Andrew Colin wrote: Normally you should open ports 1-2 udp On 9/13/2013 11:37 AM, Jonas Kellens wrote: I now see that an IP-address gets blocked by my firewall because there are packets coming onto port 11955. Why do I need such a big range ? That's like

Re: [asterisk-users] RTP port ranges

2013-09-13 Thread Andrew Colin
Because normally it will use a random port between them On 9/13/2013 11:43 AM, Jonas Kellens wrote: On 09/13/2013 11:41 AM, Andrew Colin wrote: Normally you should open ports 1-2 udp On 9/13/2013 11:37 AM, Jonas Kellens wrote: I now see that an IP-address gets blocked by my firewall

Re: [asterisk-users] RTP port ranges

2013-09-13 Thread Jonas Kellens
Hello, and when I define 11500 - 11954 it should use a random port in this range. Where is it stated that you MUST use 1-2 ??? Someone else please ? Jonas. On 09/13/2013 11:46 AM, Andrew Colin wrote: Because normally it will use a random port between them On 9/13/2013 11:43 AM,

Re: [asterisk-users] RTP port ranges

2013-09-13 Thread Johann Steinwendtner
Maybe you should open 11955 on you fw as well. This could be the rtcp port. Regards Hans On 2013-09-13 11:49, Jonas Kellens wrote: Hello, and when I define 11500 - 11954 it should use a random port in this range. Where is it stated that you MUST use 1-2 ??? Someone else please ?

Re: [asterisk-users] RTP port ranges

2013-09-13 Thread Jonas Kellens
Could be... is there no way to be sure ? Is there no way to calculate this ? Thanks, Jonas. On 09/13/2013 12:11 PM, Johann Steinwendtner wrote: Maybe you should open 11955 on you fw as well. This could be the rtcp port. Regards Hans On 2013-09-13 11:49, Jonas Kellens wrote: Hello, and

Re: [asterisk-users] RTP port ranges

2013-09-13 Thread Tony Mountifield
In article 5232dcbc.20...@telenet.be, Jonas Kellens jonas.kell...@telenet.be wrote: I have defined that I want to receive audio (RTP) on port 11500 till 11954 (rtp.conf). The same range I have defined in my firewall. I now see that an IP-address gets blocked by my firewall because there

Re: [asterisk-users] RTP port ranges

2013-09-13 Thread A J Stiles
On Friday 13 September 2013, Jonas Kellens wrote: On 09/13/2013 11:41 AM, Andrew Colin wrote: Normally you should open ports 1-2 udp On 9/13/2013 11:37 AM, Jonas Kellens wrote: I now see that an IP-address gets blocked by my firewall because there are packets coming onto port

[asterisk-users] executing the h extension at the real hangup of the call

2013-09-13 Thread Henrik Westerberg
Hi, I am running Asterisk 11.3 with both SIP and ISDN. When dialing out (always over SIP) I want to keep track of who answered and of the length of the call. [outgoing-dev2] exten = h,1,Agi(agi://localhost/ajpbxtest.agi?status=finished) exten = _X.,1,NoOp(Will send call to ${CC_DIALSTRING})

[asterisk-users] ANN: Obelus, a Python AMI/AGI library

2013-09-13 Thread Antoine Pitrou
Hello, I'm pleased to announce the first release of Obelus, a MIT-licensed Python library to interact with Asterisk using the AMI and AGI protocols. Compared to existing libraries, Obelus is framework- and programming-style-agnostic, and compatible with Python 3 as well as Python 2. It also has

Re: [asterisk-users] executing the h extension at the real hangup of the call

2013-09-13 Thread Gareth Blades
On 13/09/13 12:31, Henrik Westerberg wrote: Hi, I am running Asterisk 11.3 with both SIP and ISDN. When dialing out (always over SIP) I want to keep track of who answered and of the length of the call. [outgoing-dev2] exten = h,1,Agi(agi://localhost/ajpbxtest.agi?status=finished) exten =

Re: [asterisk-users] RTP port ranges

2013-09-13 Thread Steven Howes
On 13 Sep 2013, at 11:44, A J Stiles wrote: In the Windows world, where you usually don't get the Source Code, you never know what is running on your computer; in which case, you are never sure that there isn't a daemon listening on a particular port number, so it is wise in that case not

Re: [asterisk-users] executing the h extension at the real hangup of the call

2013-09-13 Thread jg
Is there a special reason why you do not evaluate the CDRs? The Call Detail Records would answer your questions and you could even add custom fields. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] Grnvoip

2013-09-13 Thread Mike Diehl
Does anyone know if Grnvoip is still in business, or what's going on with them? I had an account with them, but they no longer terminate calls. Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Transfer Fraud

2013-09-13 Thread Adrian Serafini
On 09/13/2013 04:12 PM, jg wrote: Is there a general recipe to avoid fraudulent calls under the following conditions? A receptionist transfers calls as a callee (customers are calling) and as a caller (boss asks to call and then transfer to him), i.e. the Dial cmd for the internal context

[asterisk-users] Transfer Fraud

2013-09-13 Thread jg
Is there a general recipe to avoid fraudulent calls under the following conditions? A receptionist transfers calls as a callee (customers are calling) and as a caller (boss asks to call and then transfer to him), i.e. the Dial cmd for the internal context contains Tt. Then an outside call

[asterisk-users] AUTO: Chris Douglas is out of the office (returning 09/16/2013)

2013-09-13 Thread Chris Douglas
I am out of the office until 09/16/2013. I will be out of the office and will have minimal access to email and voicemail. If you need immediate assistance, please contact the Pioneer I.S. Help Desk at 316-688-8777, 800-613-9382, or via intercompany dialing using the internal directory at

Re: [asterisk-users] Transfer Fraud

2013-09-13 Thread jg
create a separate context for outbound calls. Wouldn't that be more or less identical to my way? I would have to dispatch the channel to see whether it is allowed to enter the outbound context. Maybe I misunderstood something. jg --

Re: [asterisk-users] Transfer Fraud

2013-09-13 Thread Eric Wieling
This is one of the disadvantages of using phones without a transfer button. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jg Sent: Friday, September 13, 2013 4:52 PM To: adrian-li...@wombit.com; Asterisk