[asterisk-users] RTP port ranges
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 11955 when I clearly define in my SDP-body that I want to receive audio on port range 11500 till 11954 ? What makes the client choose this port number when it is not allowed ? Kind regards, Jonas. -- _ -- 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP port ranges
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 Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP port ranges
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 for 250 concurrent calls ! Jonas. -- _ -- 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP port ranges
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 because there are packets coming onto port 11955. Why do I need such a big range ? That's like for 250 concurrent calls ! Jonas. -- _ -- 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP port ranges
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, 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 11955. Why do I need such a big range ? That's like for 250 concurrent calls ! Jonas. -- _ -- 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP port ranges
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 ? 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, 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 11955. Why do I need such a big range ? That's like for 250 concurrent calls ! Jonas. -- _ -- 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP port ranges
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 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, 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 11955. Why do I need such a big range ? That's like for 250 concurrent calls ! Jonas. -- _ -- 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP port ranges
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 are packets coming onto port 11955. How come the client sends audio on port 11955 when I clearly define in my SDP-body that I want to receive audio on port range 11500 till 11954 ? What makes the client choose this port number when it is not allowed ? An RTP connection typically uses a pair of adjacent ports. The even port for the RTP stream, and the next port up (odd) for RTCP reports. So when defining a port range, you should probably make the lower port number even and the upper port number odd. (so the default 1-2 is probably wrong too, and should be 1-1) It also means that you should allow at least twice as many ports as the number of simultaneous calls you want to handle. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP port ranges
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 11955. Why do I need such a big range ? That's like for 250 concurrent calls ! Having a port open really is not a big deal, unless there's a daemon listening on it. 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 to leave ports open unnecessarily. (Though not half as wise as just not running un-audited software in the first place .) But this is the Open Source world, and we have the advantage of knowing exactly what is running our computers. Open ports going nowhere simply are not a security concern this side of the fence. -- AJS Answers come *after* questions. -- _ -- 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] executing the h extension at the real hangup of the call
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 ${CC_DIALSTRING})
exten = _X.,n,Dial(${CC_DIALSTRING}, 60, M(uploadpeer-dev2^${CC_CALLID})em)
exten =
_X.,n,Agi(agi://localhost/ajpbxtest.agi?status=faileddialstatus=${DIALSTATUS})
The h extension is called correctly when the call comes in over IP and when I
record the call. But when the call has come in over SIP the h extension is
called directly after the call is answered so all the call gets length 0 in my
own database.
I guess that I could record the calls and throw away the recordings afterwards.
In this way the RTP would stay on the server. But is there not a cleaner way to
get Asterisk to execute the h extension (or another possibility to fix a
callback somewhere) when the the Disconnect comes in over SIP?
Regards,
Henrik
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[asterisk-users] ANN: Obelus, a Python AMI/AGI library
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 an integrated test suite. This is version 0.1, and as such some APIs are a bit draftish and not guaranteed to be stable accross future releases. Also, documentation is far from exhaustive. Quick links --- * Project page: https://pypi.python.org/pypi/obelus/ * Source code, issue tracker: https://bitbucket.org/optiflowsrd/obelus * Documentation (incomplete): https://obelus.readthedocs.org Features * Python 2 and Python 3 support. * AMI, FastAGI and Async AGI support. * Event-driven API friendly towards non-blocking (async) network programming styles. * :pep:`3156`-style protocol implementations. * Framework-agnostic. * Adapters for the `Tornado`_, `Twisted`_, `Tulip`_ network programming frameworks. * Unit-tested. Requirements * Python 2.7, 3.2 or later. Regards Antoine. -- _ -- 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] executing the h extension at the real hangup of the call
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 = _X.,1,NoOp(Will send call to ${CC_DIALSTRING})
exten = _X.,n,Dial(${CC_DIALSTRING}, 60,
M(uploadpeer-dev2^${CC_CALLID})em)
exten =
_X.,n,Agi(agi://localhost/ajpbxtest.agi?status=faileddialstatus=${DIALSTATUS})
The h extension is called correctly when the call comes in over IP and
when I record the call. But when the call has come in over SIP the h
extension is called directly after the call is answered so all the
call gets length 0 in my own database.
I guess that I could record the calls and throw away the recordings
afterwards. In this way the RTP would stay on the server. But is there
not a cleaner way to get Asterisk to execute the h extension (or
another possibility to fix a callback somewhere) when the the
Disconnect comes in over SIP?
I have no idea why you are seeing the h extension being run before the
call ends. Its not something I have ever seen happen.
Whether or not Asterisk stays in the RTP media path makes no difference
as it will always stay in the SIP signalling path and its that which
controls the call establishment and termination.
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Re: [asterisk-users] RTP port ranges
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 to leave ports open unnecessarily. (Though not half as wise as just not running un-audited software in the first place .) Netstat will tell you what's running on Windows, just like on other platforms. Steve -- _ -- 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] executing the h extension at the real hangup of the call
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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grnvoip
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 to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer Fraud
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 contains Tt. Then an outside call would operate as a Local channel in an internal context after the first transfer. If the internal context allows to dial outside, which is quite common, then this can be abused by the outside caller. An obvious solution is to disallow Local channels to call outside lines, but there are some possible side effects if Local channels are used explicitly. This would require adding a persistent channel variable (the ones with __). create a separate context for outbound calls. -- _ -- 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfer Fraud
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 would operate as a Local channel in an internal context after the first transfer. If the internal context allows to dial outside, which is quite common, then this can be abused by the outside caller. An obvious solution is to disallow Local channels to call outside lines, but there are some possible side effects if Local channels are used explicitly. This would require adding a persistent channel variable (the ones with __). I apologize if this type of question has already been asked before. jg -- _ -- 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AUTO: Chris Douglas is out of the office (returning 09/16/2013)
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 http://directory.pioneer.world. There you can leave an emergency message for the on-call technician. Otherwise I will respond as soon as I can. Thanks, Chris Douglas Technical Services Manager Pioneer Balloon Company tel. 316-688-8648 fax. 316-691-6901 Note: This is an automated response to your message asterisk-users Digest, Vol 110, Issue 14 sent on 9/13/2013 12:00:01 PM. This is the only notification you will receive while this person is away. -- _ -- 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer Fraud
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 -- _ -- 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer Fraud
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 Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfer Fraud 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 -- _ -- 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
