Re: [asterisk-users] remote UPDATE command
Thanks, I will try asterisk 1.8 tomorrow and see. Sent from my iPhone On Mar 13, 2012, at 8:24 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 03/13/2012 07:19 AM, Arstan Jusupov wrote: So since remote UPDATE is not supported, this project of mine would fail. Is that correct? If your chosen provider is going to send SIP requests to your system that are not supported, and then drop calls when those requests are rejected, then yes, you have a problem. Of course, you could always choose a different provider that implements SIP properly, or you could talk to that provider about correcting the behavior of their system. Alternatively, you could move to Asterisk 1.8, which in addition to being a supported release (unlike Asterisk 1.6.2.7, which is not supported and isn't even the most recent 1.6.2.x release) also will properly respond to UPDATE requests (although it still won't advertise that it supports them, but it does). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.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 -- _ -- 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] How can I get SIP/SDP values to a variable?
Thanks for the quick reply, I was afraid of that. Oh well. :) Rennes Neps Elion Ettevõtted AS rennes.n...@elion.ee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Tuesday, March 13, 2012 1:31 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How can I get SIP/SDP values to a variable? On 03/13/2012 05:58 AM, Rennes Neps wrote: I wonder if there's a way to read SDP values into a variable in Asterisk? I have successfully used SIP_HEADER function to get all I want out of SIP part of the message, no problem. But I would like to be able to read SDP part as well, anyone? Reason I want to to this is: testing the SIP_ALG condition in incoming invite message. Majority of cases can be detected just by comparing src ip address and via, but some devices only rewrite SDP c value for example and so on ... I haven't found any information anywhere how to achieve reading SDP with asterisk. I can use 1.8 and 10 versions. There is no mechanism in Asterisk to do this. The most practical approach would probably be to put a stateless SIP proxy in front of Asterisk and write whatever logic you like there, causing it to add one or more headers to the SIP messages that can be accessed from the dialplan in Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.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 -- _ -- 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] how to show used wrong password
hi all, have asterisk set up in combination with fail2ban. all works as expected only there is 1 extension that is trying to register with a wrong password causing fail2ban to block the IP address, normally that is ok behaviour but i have several extensions on that IP address. someone has once setup this extension as a test, but it seems to be impossible to find where it was installed (probably a softphone) and its a bit far away for me to go check it out. since there is no way to stop the troubling extension i figured i might as well let it connect by setting the wrongly used password as correct, at least in that case the other extensions won't be blocked and i can try to call that extension myself. anyway to see which wrong password is being used? much obliged, Randall -- _ -- 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] how to show used wrong password
On Tuesday 13 March 2012, Randall wrote: hi all, have asterisk set up in combination with fail2ban. all works as expected only there is 1 extension that is trying to register with a wrong password causing fail2ban to block the IP address, normally that is ok behaviour but i have several extensions on that IP address. . snip . anyway to see which wrong password is being used? tcpflow. (And don't underestimate the power of simply disconnecting things until it works . last thing you disconnected was the faulty one.) -- 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
Re: [asterisk-users] Problem with ReceiveFax
On 13/03/2012 8:10 PM, Ishfaq Malik wrote: On Tue, 2012-03-13 at 00:10 +0800, Larry Moore wrote: On 12/03/2012 10:53 PM, Ishfaq Malik wrote: Thanks for the input so far. I'm going to keep plugging away and if anyone has any insights, they will be gladly appreciated. Ish In SIP Account Configuration on Draytek; Set Voice Active Detect to Off In Phone Settings on the Draytek; Enable Symmetric RTP Check Start End RTP Ports match values set in /etc/asterisk/udptl.conf for udptlstart udptlend In /etc/asterisk/udptl.conf set; use_even_ports=yes Thanks for the above, I was hoping to have replied earlier with a success message buy alas, no joy to be had. Could I be having some sort of DTMF issue? I noticed this in amongst the console output once I set the console logging level to include dtmf [2012-03-13 12:06:39] DTMF[24784]: channel.c:3976 __ast_read: DTMF end 'f' received on SIP/588-000c, duration 0 ms [2012-03-13 12:06:39] DTMF[24784]: channel.c:4002 __ast_read: DTMF begin emulation of 'f' with duration 100 queued on SIP/588-000c [2012-03-13 12:06:39] DTMF[24784]: channel.c:4138 __ast_read: DTMF end emulation of 'f' queued on SIP/588-000c does the above look correct for an inbound fax? Thanks in advance (again!) Ish It's now time to do some debugging. I would suggest you capture packets between asterisk and peer 588 using tcpdump, make sure you enable a large enough snaplen (-s) to ensure you capture all packets in the frame. Submit your fax and upon completion of the session whether or not it is received successfully, transfer the file where you can open the captured file in Wireshark and select VoIP Calls located in the Telephony menu. You can then select the relevant line or lines in the session and click on the Flow button and review what is happening. I have a Grandstream HT-503 at the other end of an IPSEC vpn which has the FXO port connected to a PSTN line. I have configured the HT-503 to call the fax extension in the dialplan when it answers a call hence I have disabled faxdetect in the peer configuration. Looking at the Draytek manual I think this would be setup in VoIP Phone Settings by enabling Call Forwarding and setting it to Always and defining the SIP URL as fax@astersk_server_ip, assuming you have a fax extension enabled in the context of the peer. I am assuming you currently have this set to 200@astersk_server_ip. Did you disable VAD on the Draytek. I would also suggest you disable Call Waiting Call Transfer. You may also want to look at Volume Gain in case that affects the level of the signal being converted to T.38 on the Draytek. Testing by progressively decreasing the level and if that doesn't help then increasing it. Here is the peer configuration I just tested with my HT-503. T.38 is enabled in the [general] section of sip.conf [0123456789] type=peer defaultuser=0123456789 secret=you_guessed_it call-limit=2 host=dynamic disallow=g722 g726nonstandard=yes ;(this is required for Sipura and Grandstream ATAs, among others). transport=udp,tcp encryption=no directmedia=no faxdetect=no context=Fax-Test qualify=yes Good luck. Larry. -- _ -- 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] ISDN, overlap and open dialing plans
Hi, I've got the following setup: PSTN/ISDN E1- Asterisk E1- Alcatel 4400 PBX TDM phones When a TDM phone is dialing out to a national number, it seems that the PBX is using enbloc dialing. When a TDM phone is dialing out to an international number (variable length numbers), it seems that the PBX is using overlap dialing as Asterisk is currently receiving truncated numbers. What is the best way to deal with such situations ? 1. configure PSTN in enbloc dialing and tweak dialplan to mimic overlap dialing ? 2. or configure both PTSN and PBX spans in overlap mode ? Suggestions ? Regards -- _ -- 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] DAHDI_SPANCONFIG failed on span 1: No such device or address (6)
Hi all, I have problems starting dahdi. dahdi_cfg -vvv allwasy comes back with: DAHDI Tools Version - 2.2.1.1 DAHDI Version: 2.3.0.1 Echo Canceller(s): Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 02) Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 03) 3 channels to configure. DAHDI_SPANCONFIG failed on span 1: No such device or address (6) I searched the internet but could not yet find a solution. I already tried to exchange the zaphfc drivers as suggested, but they did not compile. I actually did not find a new(er) tutorial how to build an Asterisk with a HFC-S card. Any suggestions / hints / tutorials / links welcome. Do I need some special drivers in the kernel ? Modprobe ? Anything else special I need ? Thanks Rainer Details: I run Debian 6.0.4 with a fresh 2.6.35 Kernel, most drivers compiled as modules. I installed the newest Asterisk 10.2.0-rc2 and dahdi_sources and compiled them, but same happened with aptitude install dahdi on older kernel. download Asterisk - ./configure - make - make install - make samples download dahdi.tar.bz2 - make - make install dahdi_hardware: driver should be 'zaphfc' but is actually 'hfcpci' pci::00:0d.0 zaphfc+ 1397:2bd0 HFC-S ISDN BRI card -- NEU: FreePhone 3-fach-Flat mit kostenlosem Smartphone! Jetzt informieren: http://mobile.1und1.de/?ac=OM.PW.PW003K20328T7073a -- _ -- 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] Capacity of single instance of Asterisk
Hi Steve Thanks for your input. Please check my comments. I have a server with 24 cores running at 2.4ghz and 16 GB RAM. How many concurrent SIP sessions I can run from single instance of Asterisk on this server? I wish to use G711 codec with echo cancel. And all calls needs to be recorded. What kind of capacity are you looking to achieve? [Amit Patkar] Some where 2400 G.711 sessions with recording. So approx 1200 calls. From my experience, Asterisk is not really much of a RAM hog. A couple GB is good for a couple hundred simultaneous calls. With 4 'Intel(R) Xeon(TM) CPU 3.40GHz' cores, I can handle a couple hundred simultaneous non-transcoding calls with no recording on Asterisk 1.2. With 24 cores and 16 GB on tap, you will probably find other resource limitations before either CPU or RAM are a limiting factor. Personally, I'm a 'don't put all your eggs in one basket' kind of guy. Assuming a simplistic linear relationship between my host and yours, what will you do when it crashes with 1600 calls in progress? What will you do when you need to install patches or upgrade or ... I like a couple of instances of OpenSIPS in front of several Asterisk instances, even if OpenSIPS is on the same boxes as Asterisk. [Amit Patkar] I completely agree with you on distributing the load. At the same time, I am looking at juicing hardware as well. Can you share the number instead of saying couple hundreds? What will be impact on no of session when G729a is used? Significant. [Amit Patkar] Can I assume 30% reduction? Or it would be much more. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] DAHDI_SPANCONFIG failed on span 1: No such device or address (6)
This means the config file says 3 ports, but no card is detected. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of rama...@gmx.de Sent: Tuesday, March 13, 2012 10:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DAHDI_SPANCONFIG failed on span 1: No such device or address (6) Hi all, I have problems starting dahdi. dahdi_cfg -vvv allwasy comes back with: DAHDI Tools Version - 2.2.1.1 DAHDI Version: 2.3.0.1 Echo Canceller(s): Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 02) Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 03) 3 channels to configure. DAHDI_SPANCONFIG failed on span 1: No such device or address (6) I searched the internet but could not yet find a solution. I already tried to exchange the zaphfc drivers as suggested, but they did not compile. I actually did not find a new(er) tutorial how to build an Asterisk with a HFC-S card. Any suggestions / hints / tutorials / links welcome. Do I need some special drivers in the kernel ? Modprobe ? Anything else special I need ? Thanks Rainer Details: I run Debian 6.0.4 with a fresh 2.6.35 Kernel, most drivers compiled as modules. I installed the newest Asterisk 10.2.0-rc2 and dahdi_sources and compiled them, but same happened with aptitude install dahdi on older kernel. download Asterisk - ./configure - make - make install - make samples download dahdi.tar.bz2 - make - make install dahdi_hardware: driver should be 'zaphfc' but is actually 'hfcpci' pci::00:0d.0 zaphfc+ 1397:2bd0 HFC-S ISDN BRI card -- NEU: FreePhone 3-fach-Flat mit kostenlosem Smartphone! Jetzt informieren: http://mobile.1und1.de/?ac=OM.PW.PW003K20328T7073a -- _ -- 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] Capacity of single instance of Asterisk
Hi Kevin, Thank for your views. Where as no one is ready to share real numbers. I am looking at benchmarks so that I can plan for resources. Since asterisk project is active for so many years, I was expecting some published numbers. Thanks Regards, Amit Patkar On 03/12/2012 03:38 PM, Steve Edwards wrote: On Mon, 12 Mar 2012, Amit Patkar wrote: What will be impact on no of session when G729a is used? Assuming that transcoding is involved; if all the system is doing is passing through G.729A media streams, and recording them in unmixed G.729A format, there's no additional impact (the system might actually perform slightly better, as there is substantially less data being shuffled around). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.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] how to show used wrong password
On 03/13/2012 08:11 AM, A J Stiles wrote:
On Tuesday 13 March 2012, Randall wrote:
hi all,
have asterisk set up in combination with fail2ban.
all works as expected only there is 1 extension that is trying to
register with a wrong password causing fail2ban to block the IP address,
normally that is ok behaviour but i have several extensions on that IP
address.
. snip .
anyway to see which wrong password is being used?
tcpflow.
(And don't underestimate the power of simply disconnecting things until it
works . last thing you disconnected was the faulty one.)
This will not help. Assuming we are talking about a SIP REGISTER here,
the password is *not* sent in the request. Asterisk issues a challenge
including a randomly generated value (called a 'nonce'), then the UA
attempting to register responds to that challenge with an MD5 digest of
a string composed of various elements, including both the nonce and the
shared secret ('password'). Asterisk computes the same digest
internally, and if they match, then the assumption is that both ends
know the shared secret.
By their very nature, digest functions are not reversible; given the MD5
digest present in an SIP request containing an Authorization header,
there is no way to figure out what shared secret was used in the
computation of that digest. Since you know the nonce and the other
portions of the calculation, you could attempt to try various 'likely'
passwords to see if any of them result in the same digest value... this
is called the brute-force method, and it could take a *very* long time
to arrive at a shared secret that would allow the endpoint to register.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
--
_
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http://www.asterisk.org/hello
asterisk-users mailing list
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Re: [asterisk-users] DAHDI_SPANCONFIG failed on span 1: No such device or address (6)
On Tue, Mar 13, 2012 at 03:30:45PM +0100, rama...@gmx.de wrote: dahdi_hardware: driver should be 'zaphfc' but is actually 'hfcpci' pci::00:0d.0 zaphfc+ 1397:2bd0 HFC-S ISDN BRI card Looks like you may need to blacklist hfcpci in /etc/modprobe.d/blacklist.conf. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.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] Capacity of single instance of Asterisk
On 03/13/2012 09:43 AM, Amit Patkar | Avhan Technologies Pvt Ltd wrote: Thank for your views. Where as no one is ready to share real numbers. I am looking at benchmarks so that I can plan for resources. Since asterisk project is active for so many years, I was expecting some published numbers. You have completely missed the point that other posters have made already on this list. Let me try to express it another way. Let's say that you were browsing at an engine manufacturer's website, looking at V-8 gasoline engines, and you found one that you liked, that you felt had a good combination of features for your project. If you then contacted the manufacturer and asked them 'how fast can this engine make a car travel', what do you think their response would be? Asterisk is a toolkit; it can be configured an infinite number of ways. Any performance measurements that are made and published apply *only* to the specific configuration that was measured; it may or may not be possible to extrapolate those into other configurations, or higher/lower capacities. There are lots of published numbers of Asterisk being used in various ways and for different purposes; whether any of them apply to your specific project is debatable, and relying on them for your project would carry some level of risk. Whether you are willing to accept that risk or not is up to you. In your specific case, as has been mentioned already, it is extremely unlikely that your proposed hardware would have any trouble with Asterisk 1.8 handling 2,400 SIP call legs (1,200 bridged calls), with the same codec being used on both sides. When you add in transcoding, that will change the system significantly, and depending on the codecs involved, the hardware may still be able to handle the load. I know from experiments I did years ago with an 8-core Xeon machine (2nd generation Xeon, so nowhere near as powerful as modern Xeon cores) that the Digium G.729 codec (software implementation) could handle over 800 channels with Asterisk 1.4; I think it's reasonable to expect that given the hardware you've proposed, transcoding 1,200 channels between G.711 ulaw and G.729A is likely to be achievable. Recording, though, is an entirely different matter. Again, since you haven't provided specifics, let's assume you are going to record the call legs 'as is' (in their native formats, unmixed). If you had 2,400 G.711 ulaw call legs to record, some simple math says that you'd need be able to push 150 megabytes per second of data onto your filesystem, on top of all the 'normal' work that Asterisk is doing. That's rather a lot, and will require that your filesystem and disk subsystem be extremely fast and well tuned. If the call legs were all G.729A, then the amount of data to write would drop to 18.75 megabytes per second, which is achievable even with inexpensive SATA disks. If you want the calls recorded in 'mixed' form (most likely in 16-bit signed linear PCM audio, since that's the easiest format to use outside of Asterisk), you'd double the amount of data going into the filesystem (now 300 megabytes per second) *and* you'd add in the CPU consumption of having to decode the incoming media streams and mix them. For G.711 ulaw this is pretty cheap and would likely not be an issue; for G.729A it's somewhat more expensive, but still might not be a problem given the amount of CPU capacity you have proposed. Now do you understand why 'benchmarks' don't provide much value for something like Asterisk? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.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] Capacity of single instance of Asterisk
From: Kevin P. Fleming kpflem...@digium.com Sent: Tuesday, March 13, 2012 11:02 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Capacity of single instance of Asterisk On 03/13/2012 09:43 AM, Amit Patkar | Avhan Technologies Pvt Ltd wrote: Thank for your views. Where as no one is ready to share real numbers. I am looking at benchmarks so that I can plan for resources. Since asterisk project is active for so many years, I was expecting some published numbers. You have completely missed the point that other posters have made already on this list. Let me try to express it another way. Let's say that you were browsing at an engine manufacturer's website, looking at V-8 gasoline engines, and you found one that you liked, that you felt had a good combination of features for your project. If you then contacted the manufacturer and asked them 'how fast can this engine make a car travel', what do you think their response would be? Asterisk is a toolkit; it can be configured an infinite number of ways. Any performance measurements that are made and published apply *only* to the specific configuration that was measured; it may or may not be possible to extrapolate those into other configurations, or higher/lower capacities. There are lots of published numbers of Asterisk being used in various ways and for different purposes; whether any of them apply to your specific project is debatable, and relying on them for your project would carry some level of risk. Whether you are willing to accept that risk or not is up to you. In your specific case, as has been mentioned already, it is extremely unlikely that your proposed hardware would have any trouble with Asterisk 1.8 handling 2,400 SIP call legs (1,200 bridged calls), with the same codec being used on both sides. When you add in transcoding, that will change the system significantly, and depending on the codecs involved, the hardware may still be able to handle the load. I know from experiments I did years ago with an 8-core Xeon machine (2nd generation Xeon, so nowhere near as powerful as modern Xeon cores) that the Digium G.729 codec (software implementation) could handle over 800 channels with Asterisk 1.4; I think it's reasonable to expect that given the hardware you've proposed, transcoding 1,200 channels between G.711 ulaw and G.729A is likely to be achievable. Recording, though, is an entirely different matter. Again, since you haven't provided specifics, let's assume you are going to record the call legs 'as is' (in their native formats, unmixed). If you had 2,400 G.711 ulaw call legs to record, some simple math says that you'd need be able to push 150 megabytes per second of data onto your filesystem, on top of all the 'normal' work that Asterisk is doing. That's rather a lot, and will require that your filesystem and disk subsystem be extremely fast and well tuned. If the call legs were all G.729A, then the amount of data to write would drop to 18.75 megabytes per second, which is achievable even with inexpensive SATA disks. If you want the calls recorded in 'mixed' form (most likely in 16-bit signed linear PCM audio, since that's the easiest format to use outside of Asterisk), you'd double the amount of data going into the filesystem (now 300 megabytes per second) *and* you'd add in the CPU consumption of having to decode the incoming media streams and mix them. For G.711 ulaw this is pretty cheap and would likely not be an issue; for G.729A it's somewhat more expensive, but still might not be a problem given the amount of CPU capacity you have proposed. Now do you understand why 'benchmarks' don't provide much value for something like Asterisk? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- Kevin This is an extremely well stated response. Bryant -- _ -- 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] DAHDI_SPANCONFIG failed on span 1: No such device or address (6)
On Tue, Mar 13, 2012 at 03:30:45PM +0100, rama...@gmx.de wrote: Hi all, I have problems starting dahdi. dahdi_cfg -vvv allwasy comes back with: DAHDI Tools Version - 2.2.1.1 DAHDI Version: 2.3.0.1 Echo Canceller(s): Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 02) kb1? Why not mg2 (or OSLEC or whatever)? Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 03) 3 channels to configure. DAHDI_SPANCONFIG failed on span 1: No such device or address (6) What's the output of lsdahdi ? I searched the internet but could not yet find a solution. I already tried to exchange the zaphfc drivers as suggested, but they did not compile. I actually did not find a new(er) tutorial how to build an Asterisk with a HFC-S card. https://gitorious.org/dahdi-extra/dahdi-linux-extra Well, mainly useful for producing patches and such). Any suggestions / hints / tutorials / links welcome. Do I need some special drivers in the kernel ? Modprobe ? Anything else special I need ? Details: I run Debian 6.0.4 with a fresh 2.6.35 Kernel, most drivers compiled as modules. http://updates.xorcom.com/pkg-voip/ has some Squeeze backports. Though they're not kept up-to-date all the time. They include zaphfc. I installed the newest Asterisk 10.2.0-rc2 and dahdi_sources and compiled them, but same happened with aptitude install dahdi on older kernel. download Asterisk - ./configure - make - make install - make samples download dahdi.tar.bz2 - make - make install dahdi_hardware: driver should be 'zaphfc' but is actually 'hfcpci' pci::00:0d.0 zaphfc+ 1397:2bd0 HFC-S ISDN BRI card You should probably blacklist hfcpci. echo 'blacklist hfcpci' /etc/modprobe.d/WHATEVER.conf (Replace WHATEVER with whatever name). Then run 'rmmod hfcpci' once to remove that module at this time. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] DAHDI_SPANCONFIG failed on span 1: No such device or address (6)
Original-Nachricht Datum: Tue, 13 Mar 2012 17:26:43 +0200 Von: Tzafrir Cohen tzafrir.co...@xorcom.com An: asterisk-users@lists.digium.com Betreff: Re: [asterisk-users] DAHDI_SPANCONFIG failed on span 1: No such device or address (6) On Tue, Mar 13, 2012 at 03:30:45PM +0100, rama...@gmx.de wrote: Hi all, I have problems starting dahdi. dahdi_cfg -vvv allwasy comes back with: DAHDI Tools Version - 2.2.1.1 DAHDI Version: 2.3.0.1 Echo Canceller(s): Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 02) kb1? Why not mg2 (or OSLEC or whatever)? OSLEC ist planned in the future. I fist have to find where to get the sources and howto compile / load them. Hints appreciated. First wanted to get basics running. Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 03) 3 channels to configure. DAHDI_SPANCONFIG failed on span 1: No such device or address (6) What's the output of lsdahdi ? lsdahdi returns nothing, while lspci 00:0d.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02 and dahdi_hardware pci::00:0d.0 zaphfc- 1397:2bd0 HFC-S ISDN BRI card I would have expected dahdi_hardware would be closer related to all dahdi commands. I searched the internet but could not yet find a solution. I already tried to exchange the zaphfc drivers as suggested, but they did not compile. I actually did not find a new(er) tutorial how to build an Asterisk with a HFC-S card. https://gitorious.org/dahdi-extra/dahdi-linux-extra Well, mainly useful for producing patches and such). Thanks, but here I don't have any experience how / where to attach these patches. Where can I find more info about it ? Any suggestions / hints / tutorials / links welcome. Do I need some special drivers in the kernel ? Modprobe ? Anything else special I need ? Details: I run Debian 6.0.4 with a fresh 2.6.35 Kernel, most drivers compiled as modules. http://updates.xorcom.com/pkg-voip/ has some Squeeze backports. Though they're not kept up-to-date all the time. They include zaphfc. Same here, I don't have any experience how / where to attach these patches. Where can I find more info about it ? I installed the newest Asterisk 10.2.0-rc2 and dahdi_sources and compiled them, but same happened with aptitude install dahdi on older kernel. download Asterisk - ./configure - make - make install - make samples download dahdi.tar.bz2 - make - make install dahdi_hardware: driver should be 'zaphfc' but is actually 'hfcpci' pci::00:0d.0 zaphfc+ 1397:2bd0 HFC-S ISDN BRI card You should probably blacklist hfcpci. echo 'blacklist hfcpci' /etc/modprobe.d/WHATEVER.conf Removed and blacklisted with echo 'blacklist hfcpci' /etc/modprobe.d/blacklist.conf Already blacklisted for unknown reasons: blacklist hfcmulti blacklist hfc4s8s_l1 blacklist wcb4xxp Don't I need all /one of these modules ? Probably not the hfc.. ? Thanks Rainer (Replace WHATEVER with whatever name). Then run 'rmmod hfcpci' once to remove that module at this time. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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 -- NEU: FreePhone 3-fach-Flat mit kostenlosem Smartphone! Jetzt informieren: http://mobile.1und1.de/?ac=OM.PW.PW003K20328T7073a -- _ -- 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] how to show used wrong password
On 03/13/2012 02:11 PM, A J Stiles wrote: On Tuesday 13 March 2012, Randall wrote: hi all, have asterisk set up in combination with fail2ban. all works as expected only there is 1 extension that is trying to register with a wrong password causing fail2ban to block the IP address, normally that is ok behaviour but i have several extensions on that IP address. . snip . anyway to see which wrong password is being used? tcpflow. (And don't underestimate the power of simply disconnecting things until it works . last thing you disconnected was the faulty one.) Thanks will give that a try. p.s. i know the method, only problem that its a time consuming process (in this case it includes a 9000 km travel and not all equipment on that side is mine) -- _ -- 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] how to show used wrong password
On 03/13/2012 03:53 PM, Kevin P. Fleming wrote:
On 03/13/2012 08:11 AM, A J Stiles wrote:
On Tuesday 13 March 2012, Randall wrote:
hi all,
have asterisk set up in combination with fail2ban.
all works as expected only there is 1 extension that is trying to
register with a wrong password causing fail2ban to block the IP
address,
normally that is ok behaviour but i have several extensions on that IP
address.
. snip .
anyway to see which wrong password is being used?
tcpflow.
(And don't underestimate the power of simply disconnecting things
until it
works . last thing you disconnected was the faulty one.)
This will not help. Assuming we are talking about a SIP REGISTER here,
the password is *not* sent in the request. Asterisk issues a challenge
including a randomly generated value (called a 'nonce'), then the UA
attempting to register responds to that challenge with an MD5 digest
of a string composed of various elements, including both the nonce and
the shared secret ('password'). Asterisk computes the same digest
internally, and if they match, then the assumption is that both ends
know the shared secret.
By their very nature, digest functions are not reversible; given the
MD5 digest present in an SIP request containing an Authorization
header, there is no way to figure out what shared secret was used in
the computation of that digest. Since you know the nonce and the other
portions of the calculation, you could attempt to try various 'likely'
passwords to see if any of them result in the same digest value...
this is called the brute-force method, and it could take a *very* long
time to arrive at a shared secret that would allow the endpoint to
register.
confirmed,
doesn't work
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Re: [asterisk-users] how to show used wrong password
On Tuesday 13 March 2012, Kevin P. Fleming wrote:
[tcpflow] will not help. Assuming we are talking about a SIP REGISTER here,
the password is *not* sent in the request. Asterisk issues a challenge
including a randomly generated value (called a 'nonce'), then the UA
attempting to register responds to that challenge with an MD5 digest of
a string composed of various elements, including both the nonce and the
shared secret ('password'). Asterisk computes the same digest
internally, and if they match, then the assumption is that both ends
know the shared secret.
Ouch. That isn't going to be so easy to spot, then! You would have to guess
a bunch of likely passwords, fake up a challenge with some known nonce, and
compare the response against those you would expect with each of the various
possible passwords. (You've already got the Source Code to do all this, of
course.)
You'll have to try the selective unplugging method instead .
--
AJS
Answers come *after* questions.
--
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[asterisk-users] Low cost BRI gateway
Greetings list, I'm trying to source a very basic ISDN BRI - SIP gateway. Unfortunately, everything I've seen seems to want to do lots of other things - registering handsets, IVRs, voicemail, etc. I only want it to present an ISDN BRI as a SIP account - I have an asterisk server for the other stuff. :-) In any other environment I'd just use one of the Digium ISDN PCIe cards, but in this case the ISDN lines come into one building and the asterisk servers are in the other building across the road, and there's no copper link between them. Any suggestions gratefully received. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Line noise/hiss on Openvox A400P card on FXO
On 01/03/12 10:05, Sebastian Arcus wrote: I have a server with an OpenVox A400P card with 2 FXO modules on it. The internal extensions are SIP Grandstream phones. When making or receiving external calls through PSTN, there is an interrupted hissing like high pitch noise - which might go away for few seconds then start again. Just a follow up to my own post. After taking apart the * server, replacing motherboard, replacing analog card with a usb FXO card from Sangoma - none of the above have helped with the problem. However: 1. I tried disabling echo cancellation on the FXO ports in DAHDI and all of a sudden the line is much clearer. The noise is still there, but because the conversation is clearer, I could drop the gains on the FXO ports to -2db and -5db. This has reduced the background hiss to a certain extent. This was really an important lesson for me - as I would normally set the default echo cancellation (128 I think) and just leave it on - unless more was required. What I really should have done is started with no echo cancellation and just add a bit at a time - up to minimum necessary. I never realised how much harm too much echo cancellation does to the sound quality on the line. 2. I have tested the setup with an old Cisco 7940 phone. To my surprise, although the noise is still there somewhere, it is nowhere near as noticeable on the Grandstream GXP280 phone. It looks like in great part the Grandstream GXP 280 is just too sensitive to line noise - or it picks up / amplifies too much the wrong frequencies. 3. Again, using Ekiga, there is virtually no line noise. They must be using some really good algorithms which clean up the line. Maybe the above will help someone. I just have to decide now if I scrap the Grandstreams and replace them with Cisco phones - or just live with the line quality. Sebastian 1. The noise is not present when calling in between internal extensions (SIP only). 2. The noise is the same on both PSTN lines. 3. The noise is NOT present when I tried two different phones directly in the PSTN line(s) (a Philips DECT phone and a BT Converse phone) Is the noise interference actually on the line, which the phones filter out because of their better electronic design (then the OpenVox card) - or is it generated somewhere in the server or on the OpenVox card? I have tried: 1. Checking the interrupts and making sure the OpenVox card has its own IRQ. 2. Moving the card around on different PCI slots. 3. Changing the second network card with a different model (the first one is integrated in the motherboard). 4. Changing the motherboard, CPU and RAM (one motherboard AMD with Sis chipset, the other one Intel). 5. Placing ferrite cores on the phone cables. 6. Checking to see if the OpenVox card gets 1000 interrupts per second and it does. 7. Upgrading the kernel from 2.6.29 to 2.6.37 8. Ran FXO tune and made sure it starts with DAHDI 9. Disabled and enable software echo cancellation - it makes no difference. The server is virtually under no load during the tests. It does have IDE hard-drives (which apparently can cause problems) - but there is not much I can do about that. I also have a Sangoma USB FXO adapter - which I'm about to install and configure to see if it makes a difference. I would really like to figure out where is the noise coming from - as I'm going a bit in circles. If I can find out for sure that the OpenVox card is either broken or low quality - I'll just have to replace it. But I can't even figure that out for sure. The specs are: CPU: Celeron 2.4GHz Asterisk 10.1.2 Dahdi 2.6.0 Hard-drives: IDE OpenVox A400P analog card Many thanks for any advice. -- _ -- 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 -- Linux vehicle CCTV - www.open-t.co.uk/iroko -- _ -- 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] ISDN, overlap and open dialing plans (Olivier)
missed the point that other posters have made already on this list. Let me try to express it another way. Let's say that you were browsing at an engine manufacturer's website, looking at V-8 gasoline engines, and you found one that you liked, that you felt had a good combination of features for your project. If you then contacted the manufacturer and asked them 'how fast can this engine make a car travel', what do you think their response would be? Asterisk is a toolkit; it can be configured an infinite number of ways. Any performance measurements that are made and published apply *only* to the specific configuration that was measured; it may or may not be possible to extrapolate those into other configurations, or higher/lower capacities. There are lots of published numbers of Asterisk being used in various ways and for different purposes; whether any of them apply to your specific project is debatable, and relying on them for your project would carry some level of risk. Whether you are willing to accept that risk or not is up to you. In your specific case, as has been mentioned already, it is extremely unlikely that your proposed hardware would have any trouble with Asterisk 1.8 handling 2,400 SIP call legs (1,200 bridged calls), with the same codec being used on both sides. When you add in transcoding, that will change the system significantly, and depending on the codecs involved, the hardware may still be able to handle the load. I know from experiments I did years ago with an 8-core Xeon machine (2nd generation Xeon, so nowhere near as powerful as modern Xeon cores) that the Digium G.729 codec (software implementation) could handle over 800 channels with Asterisk 1.4; I think it's reasonable to expect that given the hardware you've proposed, transcoding 1,200 channels between G.711 ulaw and G.729A is likely to be achievable. Recording, though, is an entirely different matter. Again, since you haven't provided specifics, let's assume you are going to record the call legs 'as is' (in their native formats, unmixed). If you had 2,400 G.711 ulaw call legs to record, some simple math says that you'd need be able to push 150 megabytes per second of data onto your filesystem, on top of all the 'normal' work that Asterisk is doing. That's rather a lot, and will require that your filesystem and disk subsystem be extremely fast and well tuned. If the call legs were all G.729A, then the amount of data to write would drop to 18.75 megabytes per second, which is achievable even with inexpensive SATA disks. If you want the calls recorded in 'mixed' form (most likely in 16-bit signed linear PCM audio, since that's the easiest format to use outside of Asterisk), you'd double the amount of data going into the filesystem (now 300 megabytes per second) *and* you'd add in the CPU consumption of having to decode the incoming media streams and mix them. For G.711 ulaw this is pretty cheap and would likely not be an issue; for G.729A it's somewhat more expensive, but still might not be a problem given the amount of CPU capacity you have proposed. Now do you understand why 'benchmarks' don't provide much value for something like Asterisk? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- Kevin This is an extremely well stated response. Bryant -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20120313/bd147276/attachment-0001.htm -- Message: 10 Date: Tue, 13 Mar 2012 17:26:43 +0200 From: Tzafrir Cohen tzafrir.co...@xorcom.com Subject: Re: [asterisk-users] DAHDI_SPANCONFIG failed on span 1: No such device or address (6) To: asterisk-users@lists.digium.com Message-ID: 20120313152642.go4...@xorcom.com Content-Type: text/plain; charset=us-ascii On Tue, Mar 13, 2012 at 03:30:45PM +0100, rama...@gmx.de wrote: Hi all, I have problems starting dahdi. dahdi_cfg -vvv allwasy comes back with: DAHDI Tools Version - 2.2.1.1 DAHDI Version: 2.3.0.1 Echo Canceller(s): Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 02) kb1? Why not mg2 (or OSLEC or whatever)? Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 03) 3 channels to configure. DAHDI_SPANCONFIG failed on span 1: No such device or address (6) What's the output of lsdahdi ? I searched the internet but could not yet find a solution. I already tried to exchange the zaphfc drivers as suggested, but they did
Re: [asterisk-users] Capacity of single instance of Asterisk
On Tue, 13 Mar 2012, Amit Patkar | Avhan Technologies Pvt Ltd wrote: [Amit Patkar] I completely agree with you on distributing the load. At the same time, I am looking at juicing hardware as well. Can you share the number instead of saying couple hundreds? In the universe of possible configurations... This is our 'slow period,' but on my hardware, handling my application, at this moment in time, one of my hosts is handling 98 calls, has (as reported by 'top -d 30') a load average of 0.79, and CPU utilization of 2.3% user and 8.9% system. Asterisk is using 85m of virtual memory and has 35m resident. I've seen 300 calls on the same host, but that was not a limit of the host, just how many callers were using the service at that point in time. Note that my application (free chat rooms) is probably more resource intensive than your undisclosed application because all the frames from the participants have to be mixed with voodoo magic in the Zaptel driver. Also note that my application uses a bunch of AGIs. Each call invokes at least 6 AGIs -- all requiring access to a MySQL database. All the AGIs are written in C. If you can draw any conclusions from the above and relate it to your application -- congratulations :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Capacity of single instance of Asterisk
Am 13.03.2012 21:13, schrieb Steve Edwards: On Tue, 13 Mar 2012, Amit Patkar | Avhan Technologies Pvt Ltd wrote: [Amit Patkar] I completely agree with you on distributing the load. At the same time, I am looking at juicing hardware as well. Can you share the number instead of saying couple hundreds? having a nearly same hardware setup as yours (double xeon 2,3 ghz six core with hyperthread = 24 cores and 12 GB of ram) i was able to push asterisk 10 up to 13500 concurrent calls at around 1800 calls per second. but this was only sip signaling. i also done some load tests with 8000 concurrent calls doing a playback of a unique file for each call and the load was around 30 but sound quality still sounds ok. but no one will every build a single host system for such many calls, you will have only problems with it. a typical sip proxy can handle much more sip messages as asterisk and you can easy spread the load over different machines. i guess you should start to try it out what your system and your asterisk configuration can handle without problems and do some educated guesses about it. so its not about numbers cause nobody can really answer this question without trying it out and there will still be too much space for difference to give you an exactly amount. best regards stefan -- _ -- 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] DAHDI_SPANCONFIG failed on span 1: No such device or address (6)
On Tue, Mar 13, 2012 at 05:10:14PM +0100, rama...@gmx.de wrote: Original-Nachricht Datum: Tue, 13 Mar 2012 17:26:43 +0200 Von: Tzafrir Cohen tzafrir.co...@xorcom.com An: asterisk-users@lists.digium.com Betreff: Re: [asterisk-users] DAHDI_SPANCONFIG failed on span 1: No such device or address (6) On Tue, Mar 13, 2012 at 03:30:45PM +0100, rama...@gmx.de wrote: Hi all, I have problems starting dahdi. dahdi_cfg -vvv allwasy comes back with: DAHDI Tools Version - 2.2.1.1 DAHDI Version: 2.3.0.1 Echo Canceller(s): Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: kb1) (Slaves: 02) kb1? Why not mg2 (or OSLEC or whatever)? OSLEC ist planned in the future. I fist have to find where to get the sources and howto compile / load them. Hints appreciated. First wanted to get basics running. Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 03) 3 channels to configure. DAHDI_SPANCONFIG failed on span 1: No such device or address (6) What's the output of lsdahdi ? lsdahdi returns nothing, while lspci 00:0d.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02 and dahdi_hardware pci::00:0d.0 zaphfc- 1397:2bd0 HFC-S ISDN BRI card I would have expected dahdi_hardware would be closer related to all dahdi commands. '-' means that no module is actually loaded. Do you have zaphfc ? If so: try: modprobe zaphfc I searched the internet but could not yet find a solution. I already tried to exchange the zaphfc drivers as suggested, but they did not compile. I actually did not find a new(er) tutorial how to build an Asterisk with a HFC-S card. https://gitorious.org/dahdi-extra/dahdi-linux-extra Well, mainly useful for producing patches and such). Thanks, but here I don't have any experience how / where to attach these patches. Where can I find more info about it ? Any suggestions / hints / tutorials / links welcome. Do I need some special drivers in the kernel ? Modprobe ? Anything else special I need ? Details: I run Debian 6.0.4 with a fresh 2.6.35 Kernel, most drivers compiled as modules. http://updates.xorcom.com/pkg-voip/ has some Squeeze backports. Though they're not kept up-to-date all the time. They include zaphfc. Same here, I don't have any experience how / where to attach these patches. Where can I find more info about it ? The -source is already patched. There is actually one big patch there (with the external drivers) and as it mentions - it is taken from the git repo I mentioned. I installed the newest Asterisk 10.2.0-rc2 and dahdi_sources and compiled them, but same happened with aptitude install dahdi on older kernel. download Asterisk - ./configure - make - make install - make samples download dahdi.tar.bz2 - make - make install dahdi_hardware: driver should be 'zaphfc' but is actually 'hfcpci' pci::00:0d.0 zaphfc+ 1397:2bd0 HFC-S ISDN BRI card You should probably blacklist hfcpci. echo 'blacklist hfcpci' /etc/modprobe.d/WHATEVER.conf Removed and blacklisted with echo 'blacklist hfcpci' /etc/modprobe.d/blacklist.conf Already blacklisted for unknown reasons: blacklist hfcmulti blacklist hfc4s8s_l1 blacklist wcb4xxp Don't I need all /one of these modules ? Those three are for a different card, so they are irrelevant (harmless, though) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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 to fax
So I'm still trying to get this to work... (I'm top posting, but the details are below, if you want/need background info) I'd like Asterisk to detect incoming faxes and redirect them elsewhere. The details aren't important, as long as I get the detection working. I've added this to my sip.conf file. Probably overkill, but I'll tune it once it works: [general] faxdetect=both My sip registrations are all in a Mysql RT database, so I added this column to my table: faxdetect char(3) default 'no' I've set faxdetect to 'yes' for the devices that I expect to be receiving fax calls. I did a sip reload from the console after adding and updating this column. I've added a fax extension to the appropriate context in extensions.conf: exten = fax,1,noop(I hear a fax!) Since most of my dialplan is in an AGI script, I've added this to the code that handles my test number: $main::agi-answer(); $main::agi-exec(ringing); $main::agi-exec(wait,5); So, now that all of this is in place, I call the extension from my fax machine... and I don't get any indication on the console that Asterisk heard a fax. My extension simply rings and I answer it. What am missing? TIA, Mike Diehl. On Friday 24 February 2012 4:22:07 pm Kevin P. Fleming wrote: On 02/24/2012 05:20 PM, Mike Diehl wrote: On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote: On 02/24/2012 05:00 PM, Mike Diehl wrote: On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote: On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote: On 02/24/2012 03:32 PM, Mike Diehl wrote: Hi all, I've got a user that has one phone number an wants to be able to us it for both voice and fax. When a fax call comes in, he wants to do some incantation on the keypad and have the call go to the fax machine. As I see it, he has 3 options. 1. (blind?) Transfer it to the fax extension. 2. Use features.conf to create a key sequence, say *2, to dial/transfer to a fax extension. 3. Use fax detect (SIP) to do it automatically. However I'm also using FFA, so I suspect these are mutually exclusive. They are not. Enabling faxdetect should do exactly what you want; it will redirect the call to the 'fax' extension in the current context, and you can then Dial() your FAX machine (or send the call to ReceiveFAX). Thank you. Then, that's what I'll do. On second though, I think my suggestion that FFA and fax detect were mutually exclusive stemmed from the idea that a call that was being originated/answered/handled by FFA would have it's call disconnected and redirected by fax detect. If this is the case, it changes my dial plan logic, and I'm not sure I fully understand what changes I'll need to make. For all I know, it might even simplify things by isolating all fax handling in one block. Well, first you should not have faxdetect enabled on outbound channels. That takes care of the 'originating' part. If you have an inbound channel that you *know* you are sending to ReceiveFAX, then you can just disable faxdetect on that channel before doing so (this is why we made 'faxdetect' configurable from the dialplan). Alternatively, you can just let calls that you *know* are going to go to ReceiveFAX (dedicated FAX DIDs, for example) just 'idle' in the dialplan listening to silence until faxdetect kicks in and sends them to ReceiveFAX. Note that the usage of FFA is not relevant here; whether you are using Fax for Asterisk, the free version of it, or res_fax_spandsp, the behavior and scenarios would be the same. Very nice. Sounds like I need to add a faxdetect column to my SIP real-time configuration. Once I've done a sip reload or pruned/loaded my user agents, I should be good to go! Didn't know faxdetect was configurable in the dialplan... Pointer to how to do it? The CHANNEL() dialplan function with the 'faxdetect' option. Not sure which releases have it; it might only be Asterisk 10. -- Take care and have fun, Mike Diehl. -- _ -- 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 to fax
#1 you might need a progress() statement after answer #2 what does sip show peer xxx look like on this peer? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Tuesday, March 13, 2012 4:18 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Transfer to fax So I'm still trying to get this to work... (I'm top posting, but the details are below, if you want/need background info) I'd like Asterisk to detect incoming faxes and redirect them elsewhere. The details aren't important, as long as I get the detection working. I've added this to my sip.conf file. Probably overkill, but I'll tune it once it works: [general] faxdetect=both My sip registrations are all in a Mysql RT database, so I added this column to my table: faxdetect char(3) default 'no' I've set faxdetect to 'yes' for the devices that I expect to be receiving fax calls. I did a sip reload from the console after adding and updating this column. I've added a fax extension to the appropriate context in extensions.conf: exten = fax,1,noop(I hear a fax!) Since most of my dialplan is in an AGI script, I've added this to the code that handles my test number: $main::agi-answer(); $main::agi-exec(ringing); $main::agi-exec(wait,5); So, now that all of this is in place, I call the extension from my fax machine... and I don't get any indication on the console that Asterisk heard a fax. My extension simply rings and I answer it. What am missing? TIA, Mike Diehl. On Friday 24 February 2012 4:22:07 pm Kevin P. Fleming wrote: On 02/24/2012 05:20 PM, Mike Diehl wrote: On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote: On 02/24/2012 05:00 PM, Mike Diehl wrote: On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote: On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote: On 02/24/2012 03:32 PM, Mike Diehl wrote: Hi all, I've got a user that has one phone number an wants to be able to us it for both voice and fax. When a fax call comes in, he wants to do some incantation on the keypad and have the call go to the fax machine. As I see it, he has 3 options. 1. (blind?) Transfer it to the fax extension. 2. Use features.conf to create a key sequence, say *2, to dial/transfer to a fax extension. 3. Use fax detect (SIP) to do it automatically. However I'm also using FFA, so I suspect these are mutually exclusive. They are not. Enabling faxdetect should do exactly what you want; it will redirect the call to the 'fax' extension in the current context, and you can then Dial() your FAX machine (or send the call to ReceiveFAX). Thank you. Then, that's what I'll do. On second though, I think my suggestion that FFA and fax detect were mutually exclusive stemmed from the idea that a call that was being originated/answered/handled by FFA would have it's call disconnected and redirected by fax detect. If this is the case, it changes my dial plan logic, and I'm not sure I fully understand what changes I'll need to make. For all I know, it might even simplify things by isolating all fax handling in one block. Well, first you should not have faxdetect enabled on outbound channels. That takes care of the 'originating' part. If you have an inbound channel that you *know* you are sending to ReceiveFAX, then you can just disable faxdetect on that channel before doing so (this is why we made 'faxdetect' configurable from the dialplan). Alternatively, you can just let calls that you *know* are going to go to ReceiveFAX (dedicated FAX DIDs, for example) just 'idle' in the dialplan listening to silence until faxdetect kicks in and sends them to ReceiveFAX. Note that the usage of FFA is not relevant here; whether you are using Fax for Asterisk, the free version of it, or res_fax_spandsp, the behavior and scenarios would be the same. Very nice. Sounds like I need to add a faxdetect column to my SIP real-time configuration. Once I've done a sip reload or pruned/loaded my user agents, I should be good to go! Didn't know faxdetect was configurable in the dialplan... Pointer to how to do it? The CHANNEL() dialplan function with the 'faxdetect' option. Not sure which releases have it; it might only be Asterisk 10. -- Take care and have fun, Mike Diehl. -- _ -- 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
Re: [asterisk-users] Transfer to fax
On Tuesday 13 March 2012 3:21:58 pm Danny Nicholas wrote: #1 you might need a progress() statement after answer I'll try that. Thank you. #2 what does sip show peer xxx look like on this peer? I'm testing against my office phone, a Polycom 501: * Name : 0004F211F1D0-2 Realtime peer: Yes, cached Secret : Set MD5Secret: Not set Remote Secret: Not set Context : customers Subscr.Cont. : Not set Language : Accountcode : 1 AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : 7001@context VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : Yes Callerid : Mike Diehl 5051234567 MaxCallBR: 384 kbps Expire : 172 Insecure : no Nat : Always ACL : Yes T.38 support : Yes T.38 EC mode : FEC T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: Yes Text Support : No Ign SDP ver : Yes Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr-IP : 173.10.242.192 Port 1811 Defaddr-IP : 0.0.0.0 Port 5060 Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 0004F211F1D0-2 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw:20) Auto-Framing : No 100 on REG : Yes Status : OK (88 ms) Useragent: PolycomSoundPointIP-SPIP_501-UA/3.1.4.0070 Reg. Contact : sip:0004F211F1D0-2@10.0.1.81 Qualify Freq : 6 ms Variables: line_id = 0004F211F1D0-2 Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs Parkinglot : -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Tuesday, March 13, 2012 4:18 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Transfer to fax So I'm still trying to get this to work... (I'm top posting, but the details are below, if you want/need background info) I'd like Asterisk to detect incoming faxes and redirect them elsewhere. The details aren't important, as long as I get the detection working. I've added this to my sip.conf file. Probably overkill, but I'll tune it once it works: [general] faxdetect=both My sip registrations are all in a Mysql RT database, so I added this column to my table: faxdetect char(3) default 'no' I've set faxdetect to 'yes' for the devices that I expect to be receiving fax calls. I did a sip reload from the console after adding and updating this column. I've added a fax extension to the appropriate context in extensions.conf: exten = fax,1,noop(I hear a fax!) Since most of my dialplan is in an AGI script, I've added this to the code that handles my test number: $main::agi-answer(); $main::agi-exec(ringing); $main::agi-exec(wait,5); So, now that all of this is in place, I call the extension from my fax machine... and I don't get any indication on the console that Asterisk heard a fax. My extension simply rings and I answer it. What am missing? TIA, Mike Diehl. On Friday 24 February 2012 4:22:07 pm Kevin P. Fleming wrote: On 02/24/2012 05:20 PM, Mike Diehl wrote: On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote: On 02/24/2012 05:00 PM, Mike Diehl wrote: On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote: On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote: On 02/24/2012 03:32 PM, Mike Diehl wrote: Hi all, I've got a user that has one phone number an wants to be able to us it for both voice and fax. When a fax call comes in, he wants to do some incantation on the keypad and have the call go to the fax machine. As I see it, he has 3 options. 1. (blind?) Transfer it to the fax extension. 2. Use features.conf to create a key sequence, say *2, to dial/transfer to a fax extension. 3. Use fax detect (SIP) to do it automatically. However I'm also using FFA, so I suspect these are mutually exclusive. They are not. Enabling faxdetect should do exactly what you want; it will redirect the call to the 'fax' extension in the current context, and you can then Dial() your FAX machine (or send the call to ReceiveFAX). Thank you. Then, that's what I'll do. On second though, I think my suggestion that FFA and fax detect were mutually exclusive stemmed from the idea that a call that was being originated/answered/handled by FFA would have it's call disconnected and redirected by fax detect. If this is the case, it changes my dial plan logic, and I'm not sure I fully understand what changes I'll need to make. For all I know, it might
Re: [asterisk-users] Transfer to fax
On 03/13/2012 04:18 PM, Mike Diehl wrote: So I'm still trying to get this to work... (I'm top posting, but the details are below, if you want/need background info) I'd like Asterisk to detect incoming faxes and redirect them elsewhere. The details aren't important, as long as I get the detection working. I've added this to my sip.conf file. Probably overkill, but I'll tune it once it works: [general] faxdetect=both This will have no effect; see below. My sip registrations are all in a Mysql RT database, so I added this column to my table: faxdetect char(3) default 'no' I've set faxdetect to 'yes' for the devices that I expect to be receiving fax calls. 'faxdetect' is not a chan_sip configuration option (unlike chan_dahdi). It's a feature that can be enabled on a channel via the CHANNEL() dialplan function. In the dialplan itself, you'd use something like this: exten = 1234,5,Set(CHANNEL(faxdetect)=yes) To do this in a configuration file, so that it will be applied to channels as soon as they are created, use 'setvar': [peer1] setvar=CHANNEL(faxdetect)=yes I'm not sure how this would be done using Realtime configuration, but it should be possible. I'd encourage you to test it out using a non-Realtime peer first, just to make sure that it works the way you expect. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.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
[asterisk-users] Getting Mac Address on connected IP phones
I am struggling to get the mac-addresses of IP phones that are connected to asterisk as the phone are in different VLAN with * and they were manually configured. I want to centralize their configuration using res_phoneprov or tftp I have tried nmap and arp in vain. Any idea? Sam -- _ -- 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] Getting Mac Address on connected IP phones
Ping the phones, then run arp. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of resea...@businesstz.com Sent: Tuesday, March 13, 2012 4:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Getting Mac Address on connected IP phones I am struggling to get the mac-addresses of IP phones that are connected to asterisk as the phone are in different VLAN with * and they were manually configured. I want to centralize their configuration using res_phoneprov or tftp I have tried nmap and arp in vain. Any idea? Sam -- _ -- 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] Transfer to fax
On Tuesday 13 March 2012 3:45:18 pm Kevin P. Fleming wrote: On 03/13/2012 04:18 PM, Mike Diehl wrote: I've set faxdetect to 'yes' for the devices that I expect to be receiving fax calls. 'faxdetect' is not a chan_sip configuration option (unlike chan_dahdi). It's a feature that can be enabled on a channel via the CHANNEL() dialplan function. In the dialplan itself, you'd use something like this: exten = 1234,5,Set(CHANNEL(faxdetect)=yes) This function was implemented somewhere in the 10.x code base, I believe. I'm running 1.6.x. So, it sounds like I need to plan an upgrade in order to get this to work. To do this in a configuration file, so that it will be applied to channels as soon as they are created, use 'setvar': [peer1] setvar=CHANNEL(faxdetect)=yes I'm not sure how this would be done using Realtime configuration, but it should be possible. I'd encourage you to test it out using a non-Realtime peer first, just to make sure that it works the way you expect. I've used setvar in my RT config and it works very well. WRT the upgrade, I've gone in and made some code changes to the voicemail module which I'll have to port over to version 10.x. Sounds like I should sign up to be a developer so I can pass those patches on... -- Take care and have fun, Mike Diehl. -- _ -- 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 to fax
On 14/03/2012 5:18 AM, Mike Diehl wrote: So I'm still trying to get this to work... (I'm top posting, but the details are below, if you want/need background info) I'd like Asterisk to detect incoming faxes and redirect them elsewhere. The details aren't important, as long as I get the detection working. I've added this to my sip.conf file. Probably overkill, but I'll tune it once it works: [general] faxdetect=both My sip registrations are all in a Mysql RT database, so I added this column to my table: faxdetect char(3) default 'no' I've set faxdetect to 'yes' for the devices that I expect to be receiving fax calls. I did a sip reload from the console after adding and updating this column. I've added a fax extension to the appropriate context in extensions.conf: exten = fax,1,noop(I hear a fax!) Since most of my dialplan is in an AGI script, I've added this to the code that handles my test number: $main::agi-answer(); $main::agi-exec(ringing); $main::agi-exec(wait,5); So, now that all of this is in place, I call the extension from my fax machine... and I don't get any indication on the console that Asterisk heard a fax. My extension simply rings and I answer it. What am missing? In your peer config set directmedia=no and faxdetect=cng, Asterisk needs to be in the path to hear the CNG tones. Larry. -- _ -- 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 to fax
On 03/13/2012 04:56 PM, Mike Diehl wrote: On Tuesday 13 March 2012 3:45:18 pm Kevin P. Fleming wrote: On 03/13/2012 04:18 PM, Mike Diehl wrote: I've set faxdetect to 'yes' for the devices that I expect to be receiving fax calls. 'faxdetect' is not a chan_sip configuration option (unlike chan_dahdi). It's a feature that can be enabled on a channel via the CHANNEL() dialplan function. In the dialplan itself, you'd use something like this: exten = 1234,5,Set(CHANNEL(faxdetect)=yes) This function was implemented somewhere in the 10.x code base, I believe. I'm running 1.6.x. So, it sounds like I need to plan an upgrade in order to get this to work. Right, so prior to that version, the *only* channel driver that had 'faxdetect' functionality was chan_dahdi. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.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] how to show used wrong password
Ouch. That isn't going to be so easy to spot, then! You would have to guess a bunch of likely passwords, fake up a challenge with some known nonce, and compare the response against those you would expect with each of the various possible passwords. (You've already got the Source Code to do all this, of course.) You'll have to try the selective unplugging method instead . There may be a way to do this, even in the face of the nonce-and-hash security system. As I understand it: when a system (re)registers with a good password, what you'll typically see is: - A registration request from the client (with the client's ID in the SIP parameters) - A response from Asterisk, saying something on the order of Stale authentication. Try again. Here's a new nonce for you. - Another registration request from the same client, specifying the newly-issued nonce, and having a hash based on that nonce and the shared secret. - An OK response from Asterisk. When a system (re)registers, and has the wrong password/secret, the exchange will be different. - A registration request from the client (with the client's ID in the SIP parameters) - A response from Asterisk, saying something on the order of Stale authentication. Try again. Here's a new nonce for you. - Another registration request from the same client, specifying the newly-issued nonce, and having a hash based on that nonce and the shared secret. - A response from Asterisk, rejecting the second registration request with something like a bad digest error. So, if you examine all of the SIP protocol exchanges taking place, you should see a whole bunch of successful four-way handshakes (from clients that have the correct secrets), and an occasional four-way handshake failure (from the one client that has the wrong password in its configuration). You won't be able to tell what password the client is actually trying to use - that's the whole point of the nonce-and-hash approach - but you'll be able to identify its client name, and (unless the far end is using a NAT or proxy) its IP address. To pin down the actual location of the client, you'll either have to go there, or have someone at the remote site do some investigation and (possibly) packet tracing on the LAN. Or, I suppose one could simply use Asterisk to try to phone the device or softphone in question, at whatever address it called in from, and ask whoever answers the phone to disable it! -- _ -- 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 to fax
On Tuesday 13 March 2012 4:04:31 pm Kevin P. Fleming wrote: On 03/13/2012 04:56 PM, Mike Diehl wrote: On Tuesday 13 March 2012 3:45:18 pm Kevin P. Fleming wrote: On 03/13/2012 04:18 PM, Mike Diehl wrote: I've set faxdetect to 'yes' for the devices that I expect to be receiving fax calls. 'faxdetect' is not a chan_sip configuration option (unlike chan_dahdi). It's a feature that can be enabled on a channel via the CHANNEL() dialplan function. In the dialplan itself, you'd use something like this: exten = 1234,5,Set(CHANNEL(faxdetect)=yes) This function was implemented somewhere in the 10.x code base, I believe. I'm running 1.6.x. So, it sounds like I need to plan an upgrade in order to get this to work. Right, so prior to that version, the *only* channel driver that had 'faxdetect' functionality was chan_dahdi. So, I have a few long nights ahead of me! Thanks for your time. -- Take care and have fun, Mike Diehl. -- _ -- 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] Installing Dahdi, libpri of different versions in one pc
Why would you want to even bother testing EOL products, such as 1.4x and 1.6.x.x? Although I am a 1.4 Luddite, I really don't quite understand why you can't test with 1.8.x or 10, where you mihgt have a hope of getting something fixed if there is a problem, unless you already KNOW there is an issue with later versions. JMO John Novack Gopalakrishnan N wrote: Hi, I would like to install Dahdi, libpri and Asterisk of different versions in one machine. Lets say, Asterisk 1.6.X, Asterisk 1.8.x and Asterisk 1.4.x to be installed in one machine, this can be done using prefix while building configure. For dahdi, libpri can it be done in same way? Because I need to test telephony cards (PRI, BRI, GSM Transcoding) with different versions of Asterisk, libpri and Dahdi, I can't remove and install again of each versions since it is time consuming, sicne there are lot of versions available. Any comments would be appreciated. Thanks. -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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 to fax
The faxdetect option is documented in the 1.8 sip.conf.sample. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Tuesday, March 13, 2012 6:17 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Transfer to fax On Tuesday 13 March 2012 4:04:31 pm Kevin P. Fleming wrote: On 03/13/2012 04:56 PM, Mike Diehl wrote: On Tuesday 13 March 2012 3:45:18 pm Kevin P. Fleming wrote: On 03/13/2012 04:18 PM, Mike Diehl wrote: I've set faxdetect to 'yes' for the devices that I expect to be receiving fax calls. 'faxdetect' is not a chan_sip configuration option (unlike chan_dahdi). It's a feature that can be enabled on a channel via the CHANNEL() dialplan function. In the dialplan itself, you'd use something like this: exten = 1234,5,Set(CHANNEL(faxdetect)=yes) This function was implemented somewhere in the 10.x code base, I believe. I'm running 1.6.x. So, it sounds like I need to plan an upgrade in order to get this to work. Right, so prior to that version, the *only* channel driver that had 'faxdetect' functionality was chan_dahdi. So, I have a few long nights ahead of me! Thanks for your time. -- Take care and have fun, Mike Diehl. -- _ -- 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] Transfer to fax
On 03/13/2012 05:45 PM, Eric Wieling wrote: The faxdetect option is documented in the 1.8 sip.conf.sample. Right, I forgot about that. Now I've really confused things. /me heads back to his hole -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.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] Getting Mac Address on connected IP phones
On 3/13/12 5:53 PM, Danny Nicholas wrote: Ping the phones, then run arp. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of resea...@businesstz.com Sent: Tuesday, March 13, 2012 4:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Getting Mac Address on connected IP phones I am struggling to get the mac-addresses of IP phones that are connected to asterisk as the phone are in different VLAN with * and they were manually configured. I want to centralize their configuration using res_phoneprov or tftp I have tried nmap and arp in vain. Any idea? ping + arp isn't going to work if they're on a different VLAN. I believe this will work: 1) Set up your TFTP server, but do not put any configuration files in the /tftpboot directory (or whatever the directory is). 2) Set the DHCP server on the phones' network to hand out the TFTP server address. 3) Reboot the phones 4) Watch the TFTP server logs and you should see each phone request a file based on its MAC. With no downloaded config file, the phone should revert to what it already has in nvram. 5) Collect MAC addresses out of the server logs 6) Profit? -- _ -- 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] Low cost BRI gateway
Dear Chris How about to use 2 Asterisk system interconnected through Wireless solution point to point one system should be for ISDN BRI gateway with Digium PCI card and the other server for extension voice mal and so on. for covering distances it will better to use Motorola Canopy web site or Airmax from Ubiquiti (web site http://www.ubnt.com/airmax). It should be as the following (- -) PSTN/ISDN BRI Lines -BRI- Asterisk-LAN- /\ Wireless point to point /\ -LAN Asterisk customer side registering handsets, IVRs, voicemail, etc In general these system operates on 900 MHz 2.4 GHz, 5.2 GHz, 5.4 GHz, and 5.7 GHz and can cover a long distances depending to environment. Please be advise to check local regulation about using wireless system (frequency operation range, authorization and others. Best regards Mc GRATH Ricardo E-Mail mcgra...@mail2web.com -- _ -- 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] Getting Mac Address on connected IP phones
On Mar 13, 2012, at 5:51 PM, resea...@businesstz.com wrote: I am struggling to get the mac-addresses of IP phones that are connected to asterisk as the phone are in different VLAN with * and they were manually configured. I want to centralize their configuration using res_phoneprov or tftp I have tried nmap and arp in vain. Your router knows the MAC addresses of the phones. So does your DHCP server, if they are using DHCP. -- _ -- 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] Capacity of single instance of Asterisk
On Tuesday 13 Mar 2012, Amit Patkar | Avhan Technologies Pvt Ltd wrote: Thank for your views. Where as no one is ready to share real numbers. I am looking at benchmarks so that I can plan for resources. Since asterisk project is active for so many years, I was expecting some published numbers. We're running some 400 simultaneous calls with recording and no transcoding on a 2xQuad-core Intel boxes, 16GB RAM. The box is serving SIP clients and passes calls over an IAX2 trunk to the PSTN-connected box. Load average rarely goes above 0.5. Recording is done on a RAID array attached to a separate SCSI controller, which makes a lot of difference to performance. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] Getting Mac Address on connected IP phones
James Sharp wrote: On 3/13/12 5:53 PM, Danny Nicholas wrote: Ping the phones, then run arp. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of resea...@businesstz.com Sent: Tuesday, March 13, 2012 4:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Getting Mac Address on connected IP phones I am struggling to get the mac-addresses of IP phones that are connected to asterisk as the phone are in different VLAN with * and they were manually configured. I want to centralize their configuration using res_phoneprov or tftp I have tried nmap and arp in vain. Any idea? ping + arp isn't going to work if they're on a different VLAN. I believe this will work: 1) Set up your TFTP server, but do not put any configuration files in the /tftpboot directory (or whatever the directory is). 2) Set the DHCP server on the phones' network to hand out the TFTP server address. 3) Reboot the phones 4) Watch the TFTP server logs and you should see each phone request a file based on its MAC. With no downloaded config file, the phone should revert to what it already has in nvram. 5) Collect MAC addresses out of the server logs 6) Profit? Handy but working plan. Let me give it a try Thanks Sam -- _ -- 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] Installing Dahdi, libpri of different versions in one pc
Its because the card what I have only work with 1.4 and 1.6. On Wed, Mar 14, 2012 at 4:05 AM, John Novack jnov...@stromberg-carlson.orgwrote: ** Why would you want to even bother testing EOL products, such as 1.4x and 1.6.x.x? Although I am a 1.4 Luddite, I really don't quite understand why you can't test with 1.8.x or 10, where you mihgt have a hope of getting something fixed if there is a problem, unless you already KNOW there is an issue with later versions. JMO John Novack Gopalakrishnan N wrote: Hi, I would like to install Dahdi, libpri and Asterisk of different versions in one machine. Lets say, Asterisk 1.6.X, Asterisk 1.8.x and Asterisk 1.4.x to be installed in one machine, this can be done using prefix while building configure. For dahdi, libpri can it be done in same way? Because I need to test telephony cards (PRI, BRI, GSM Transcoding) with different versions of Asterisk, libpri and Dahdi, I can't remove and install again of each versions since it is time consuming, sicne there are lot of versions available. Any comments would be appreciated. Thanks. -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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
