Re: [asterisk-users] RFC List
On 08/08/2012 06:30 AM, Kannan wrote: Where can I get a complete set of RFCs and other specifications supported by Asterisk? To my knowledge there is no such list. In addition, Asterisk (like many other pieces of software) does not claim 100% compliance with every RFC that is relevant, so usually it's better to ask about the specific features you are interested in. -- 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] So long, and thanks for all the fish!
I've been with Digium for just over seven years, and it's been an incredible experience that I wouldn't have traded for anything. When Mark Spencer invited me to visit Digium (and Huntsville) in early 2005, I could not have dreamed that I'd end up working for such an exciting, innovative company, finding a wife, and meeting hundreds of people (many of whom are now friends) around the world. It's been a time of tremendous personal and career growth, and my wonderful colleagues at Digium and in the Asterisk open source community have been directly responsible for most of that. Recently, though, I've been presented an opportunity to take on a new challenge and this has resulted in my acceptance of a new job, in a new industry. In the middle of September, I'll start working for Bloomberg, L.P., in the Office of the CTO, helping to lead their nascent open source initiative. I'll be working to bring the power of open source software, open standards, and community building to the financial market data services industry, where it is sorely needed (and overdue). Michelle and I will be relocating to the greater New York City area, but Michelle will continue in her role as Digium's in-house counsel. Because of our need to relocate, I'll only be at Digium until August 8th, although I'll be in Huntsville until around Labor Day. This is yet another incredibly exciting, career changing opportunity in my life, and I can't wait to see what it will bring. I'll be forever thankful for the opportunity that Digium and the Asterisk community provided me to learn, grow and find the place where my skills and experience are the most valuable (to both myself and my employer). -- 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] best PRI gateway?
On 07/28/2012 05:43 PM, Mike wrote: what are folks using for PRI gateways these days? Obviously there's lots of folks using TE410s and related cards, which work well, and I know reasonably well. However, anyone using anything standalone that stands out as being particularly stellar? Anything that: * takes a couple of PRIs or more (or, if it's not costly, I'm not opposed to two of 'em) * does a good job of echo cancellation (128ms? what's the standard now?) * isn't horrendously expensive * straightforward to configure * doesn't leave you annoyed every time there's a problem More looking for real world experiences rather than vendors trying to sell, of course. Devices that you've plugged in, configured, and then forgotten about (since they never give you trouble) are generally the most sysadmin-friendly! I'm clearly biased, but I suggest you consider the Digium G100/G200 gateways. We've had really good feedback from users since they were released, and they certainly meet all of your needs listed above (and there are a few more, like T.38 FAX support on all channels). -- 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] still got ReceiveFax() problem, how to properly setup asterisk fax?
On 07/26/2012 10:33 PM, Roi Stork wrote: I've posted my problem with ReceiveFax() a long time ago. Majority of the incoming faxes still end up with a T2 timeout or hangup (fax session hangup) errors. Our Setup: - we're using the Digium Free Fax module for Asterisk, all settings are default - incoming/outgoing faxes go through an E1 line - faxes are outgoing/received via Sangoma A104DE Card - fax .tiff image is converted to pdf and sent to email - clock source has already been set to NORMAL (from the E1 line), and hardware/software echo cancellation already disabled Why would you disable echo cancellers? That's a terrible idea. -- 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] still got ReceiveFax() problem, how to properly setup asterisk fax?
On 07/27/2012 09:53 AM, Eric Wieling wrote: People seem to think that Asterisk won't disable the Echo Canceler when a fax tone is detected. Why they think that is a total mystery to me,. Asterisk doesn't do this, the echo canceller itself does (or DAHDI does, in some cases). With modern ECs, as well, they don't even get disabled when a CED tone (from the answering FAX endpoint) is heard, they instead just turn off their non-linear processors, because it's been found through years of experience that leaving 'most' of the EC still in place makes FAX calls more reliable than if it was completely disabled. Since the OP has a Sangoma card with an Octasic hardware echo canceller on it, he should just leave it alone and let it do its job :-) Turning it off is probably making things worse. Sometimes I wish it was possible to selectively eradicate 'conventional wisdom' from the world! -- 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] Polycom Presence with Asterisk 1.8.12.0
On 07/26/2012 03:32 PM, Danny Nicholas wrote: Question 1 - I think asterisk only supports a limited set of statuses Asterisk does not *receive* presence updates from Polycom phones (or really, non-Digium phones) at all. Instead, the presence (status) updates you are seeing appear on your phones are the statuses that Asterisk itself generates based on the phones' activity. -- 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] Less good call quality using Asterisk
On 07/25/2012 06:42 AM, Stefan at WPF wrote: Hmm is it possible, that the monitor command changes the quality? If not I guess I also once have to try compiling it from source, though I wanted to avoid that. It certainly can, since recording the call causes disk I/O as the audio is written out. In addition, Monitor is more prone to this problem than MixMonitor is, because Monitor's call recording is done in the same thread that handles the call's audio normally. If you switch to MixMonitor, you'll probably have better results, unless your system just can't handle recording the call without overloading its CPU. -- 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] T.38 Gateway
On 07/23/2012 06:30 AM, Paul Goldbaum wrote: Hi everyone, we're trying to setup a fax gateway from PRI to SIP. Our test setup consists of two asterisk 10 machines connected over PRI and a third one which is acting as a fax endpoint over SIP. This 'third' system is the one responsible for initiating the switchover to T.38 mode for the call, since it is the last element in the path to the eventual called FAX terminal. On the machine which is receiving over PRI, we see the following message: [Jul 23 12:54:23] DEBUG[18288] res_fax.c: detected v21 preamble on SIP/xxx.xxx.xxx.xxx-0001, but DAHDI/i1/-2 does not support T.38 for T.38 gateway session This is expected behavior. If you run a gateway on a SIP channel, it will naturally assume that *that* channel is not capable of T.38 support (thus the need for a gateway) and it will attempt to request T.38 support from the channel it is bridged to. In your configuration, the Asterisk system acting as SIP FAX endpoint should *itself* request a switch to T.38 on the SIP channel before it ever emits any FAX tones at all. The system in the middle will receive this T.38 negotiation request, and the gateway code (since it has been enabled) will realize that the DAHDI channel is not T.38 capable, and it will step in to provide T.38 gateway services. -- 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] Less good call quality using Asterisk
On 07/23/2012 10:56 AM, Stefan at WPF wrote: For private use the RPI is really cool for that, though I am not yet sure if it works 100% without problems - at least it did in my latest tests. Anyone has any hint on the call quality or if Asterisk does any kind of transcoding of the audio? If both legs of the call are using the same codec, then normally Asterisk would not modify the audio in any way at all. -- 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] T.38 (PRI) Fax Debugging
On 07/20/2012 09:48 AM, Eric Wieling wrote: Neither of your questions relate to T.38 :-) Maybe you meant T.30 instead. 1) Does anyone know of any software to debug the g711cap audio files Asterisk's res_fax generates? Google has not been very helpful. That will depend a lot on what you mean by 'debug'. We have found them useful to determine whether the audio path was 'clean' or not. For example, loading them into Audacity and looking at a spectral plot will tell you whether there were any extreme spikes, which usually are indicative of packet loss. You can also use them to look at the timing of the protocol interactions (delay between transitions between sending and receiving). If your goal is to actually demodulate the audio into data and then interpret the T.30 transactions, I'm not aware of any easily available tools for doing that. Commetrex (the vendor of the FAX stack in res_fax_digium) has one that they use for helping to analyze problems, but I don't believe they make it available outside their company. There might be something in spandsp, but I haven't looked. 2) These files are in WAV format, but my Windows Media Player cannot play them. The Linux file command reports RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 8000 Hz Does anyone have a solution? Is it a bug? I have never attempted to open them with Windows Media Player, so can't be of much help there. I know that Audacity will open them and play the audio properly, because that's what we used when we developed this 'audio capture' feature. -- 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] Voicemail Emails
On 07/20/2012 04:35 PM, Josh Hopkins wrote: Yes and you just get html code in the email rather than the html format. app_voicemail does not send MIME-encapsulated emails, it sends raw email. Unless you could add a Content-Type header to the message (which app_voicemail doesn't allow you to do), the email client that receives the HTML is going to treat it as plain text, which is what you saw. -- 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] Voicemail Emails
On 07/20/2012 04:48 PM, Warren Selby wrote: On Fri, Jul 20, 2012 at 12:53 PM, Josh Hopkins j...@prorivertech.com mailto:j...@prorivertech.com wrote: Has anyone been able to make an html template for the voicemail emails. We would love to customize them beyond just plain text. We have dome some Google searches and have not been able to come up with much. __ __ I believe that Switchvox has customized the voicemail email into html. Has anyone ever tried this? Thanks, /Josh What about changing 'mailcmd=' to a shell script that rewrites the email in the format you want before sending it to sendmail? That is most likely the best way to accomplish it; it would need to receive the already-composed email on stdin, then parse it, modify it, and regenerate it before sending it to the real mailer. This could be done using standard email libraries in many scripting languages. -- 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] Channel is rsrvd and does not turn off
On 07/19/2012 03:49 PM, Rodrigo Lang wrote: I tried to shut down the channels with the command channel request hangup Khomp_SMS/B0C2-0 (Khomp_SMS/B0C2-0 the channel is locked), but nothing happens. I can only release these channels when I restart asterisk. You will probably need to ask the person(s) who made the channel driver you are using, since it's not part of Asterisk itself. -- 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] Using Asterisk 10.6 as a T38 Fax gateway
On 07/18/2012 06:30 AM, Alejandro Recarey wrote:
Hi all, and thanks for taking the time to read this.
I am trying to configure Asterisk 10.6 as a T38 Fax gateway. I am
receiving calls through the PSTN and want to send them to my VoIP
carriers as T38. This is my dialplan:
[fax]
exten = _X.,1,Set(FAXOPT(t38gateway)=yes,20)
exten = _X.,n,Dial(SIP/${EXTEN}@x.x.x.x)
I have tried with both FAXOPT(t38gateway) and FAXOPT(gateway). I have
also tried setting t38pt_udptl = yes,redundancy in sip.conf. None of
these things work. When we send a fax:
You say they don't work, but you don't provide any details (console
output, log messages, etc.) The configuration you have provided above is
*required* for T.38 support and T.38 gateway mode. If it's not working,
we are going to need more details about what is actually happening (if
anything is at all).
1. Asterisk does NOT send a REINVITE with the t38 offered. Reading the
documentation, it should detect the fax tone with the audiohook and
then send a REINVITE with t38 capability.
This is expected behavior. Proper implementations of T.38 require that
the gateway in front of the *called* endpoint monitor for FAX tones and
initiate the switch to T.38 mode. In your configuration, that would be
your carrier's gateway, assuming it is terminating the call to a
non-T.38 endpoint. If your carrier is handing off the call to another
SIP provider, then the responsibility lies with them, and so on.
However, Asterisk's T.38 gateway functionality should still detect the
V.21 preamble generated by the called FAX endpoint and initiate a switch
to T.38, if the carrier does not do it first. If this is not happening,
we'll need to see logs and console output to figure out why. What codec
are you using for your SIP calls?
2. Asterisk does not offer t38 in the SDP of the initial INVITE. This
is not a problem if it correctly detects and REINVITES for faxes, but
our destination carriers tell us that they cannot do the REINVITE
themselves because we do not offer t38 in our SDP, so they believe we
do not have that capability.
This is bizarre; there is no specification anywhere that would indicate
that a carrier should do this, and there are plenty of documents
describing how it is a *bad* idea to offer a second media stream for
T.38 in the initial INVITE of a call. I would urge you to ask them to
reconsider this behavior.
Obviously I would prefer to just detect the fax myself and have
asterisk do the REINVITE.
This is not as reliable as the far-end gateway doing it, especially if
the codec in use for the VoIP leg(s) of the call distorts the V.21
preamble in any significant way.
I have read all of the documentation on the asterisk wiki (which is
rather short) and anything else I could find online. Unfortunately
most of it is out of date and refers to asterisk versions 1.4 to 1.8,
which do not have T38 Gateway capability.
The documentation on the wiki is short, but it's complete. Enabling T.38
gateway functionality in Asterisk 10 is in fact pretty simple :-)
Problems arise, as they always do in T.38-land, because no two T.38
implementations are the same, and the choices made by carriers,
gateway/softswitch/SBC manufacturers, and others, result in
interoperability problems.
--
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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Re: [asterisk-users] Asterisk 1.8.13 / res_fax / res_fax_digium
On 07/18/2012 10:06 AM, Eric Wieling wrote: We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13 The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf indicate v34 is supported, but when I enable it I get the message res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored. Is v34 only supported with SpanDSP? Those docs are in error. V.34 is not supported. I'll notify our documentation people. Thanks for the report. -- 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] Asterisk 1.8.13 / res_fax / res_fax_digium
On 07/18/2012 10:51 AM, Eric Wieling wrote: Thank you. While you are at it, ask them to document where the audio / data from fax set g711cap| t38cap on is saved to. 8-) That is documented in the CLI help for the commands themselves; the capture files are placed into subdirectories of the main Asterisk log directory. -- 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] chan_sip sending from wrong source address when multiple interfaces are used
On 07/12/2012 09:19 AM, Benny Amorsen wrote: Kevin P. Fleming kpflem...@digium.com writes: That's quite interesting; can you describe a scenario where this occurs? Imagine you have a server with two interfaces, eth0 with 192.168.1.1/24 and eth1 with 10.0.2.1/24. Further imagine that you wish to be able to move phones between the networks without changing the SIP server address, so you set 192.168.1.1 as the SIP server no matter which network they happen to be on. Now the phones which happen to be connected to eth1 will send a request to 192.168.1.1. If Asterisk is bound to 0.0.0.0, the reply will come from 10.0.2.1. This could be solved if Asterisk did a connect() to the socket and use the same socket for answering. That would tell the system IP stack that this is in fact a connection, and so the system would ensure that the reply source IP would be correct. I must be missing something. If a phone sends a UDP packet to 192.168.1.1, how does that get routed to (arrive at) the 10.0.2.1 interface on the Asterisk server? The only way I can imagine that happening is if a router in between the phone and the server has been told that 192.168.1.0/24 is reachable *through* 10.0.2.1, which seems like a bizarre way to construct a network. Getting replies from Asterisk *back* to the phone would also require the IP stack on the Asterisk server to route those replies back over the 10.0.2.0/24 interface instead of the 192.168.1.0/24, which doesn't make any sense either. chan_sip does have the ability to use connect()-ed sockets for dialogs now, since that is required for TCP, TLS and WebSocket support. It wouldn't be a huge leap to use them for UDP as well, if that was beneficial. -- 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] chan_sip sending from wrong source address when multiple interfaces are used
On 07/11/2012 11:36 PM, Jeff LaCoursiere wrote:
This does exhibit the problem though. Your OS stack assumes one of
those addresses - the first identified interface? - is the one that all
replies will appear to come from. So phones on the 192.168.2.0/24
network that try to register get replies from 192.168.1.1 and ignore
them.
No, I don't think it does. If the server has four interfaces, on subnets
192.168.{1,2,3,4}.0/24, those are *not* overlapping, and everything will
work as expected. If a UDP packet is received on the third interface,
from an address reachable via routes over that interface, then the reply
to that packet will be sent out over that same interface, with the
source address set to the address assigned to that interface. Servers
are setup this way all the time, and it works as it should.
There must be more to the network configuration than something this
simple in order to cause the IP stack on the Asterisk server to choose
the wrong source IP address for outbound packets.
--
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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Re: [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used
On 07/12/2012 12:38 PM, Freddi Hansen wrote: We have since Asterisk 1.2 been using a configuration with 6 NIC's bonding to 3 networks, one public internet and 2 private networks. Routing calls between networks and having phones on all 3 networks is no problem. There is one case though where we do fixup with iptables. We have 30 virtuel adresses on one of the private networks and when Asterisk sends a packet to a destination then the first address of the NIC is inserted as source by the OS. example one NIC has ip's 192.168.0.10,192.168.0.20,192.168.30 Telephone (192.168.0.100) sends a packet to Asterisk 192.168.0.30, Asterisk sends response to 192.168.0.100 but with source address 192.168.0.10 as thats the first ip on that NIC. In Iptables OUTPUT q we do a set-mark to an index into our source ip's then in POSTROUTING we insert the source adr using the mark Yes, this is the situation I referred to earlier. In your case, it's all on one interface, but the server has multiple addresses on the *same* network, and thus it cannot know (without help) with address should be used for outbound packets. -- 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] chan_sip sending from wrong source address when multiple interfaces are used
On 07/12/2012 03:53 PM, Benny Amorsen wrote: chan_sip does have the ability to use connect()-ed sockets for dialogs now, since that is required for TCP, TLS and WebSocket support. It wouldn't be a huge leap to use them for UDP as well, if that was beneficial. It would be greatly appreciated :) It is low priority for the Asterisk project, as there are always workarounds. I've just looked into this a bit, and I don't see how using connect() would actually solve the problem. If we receive a UDP datagram from a SIP endpoint, we could use socket() and connect() to create a socket specifically for sending to (and receiving from) that endpoint in the future, but we can't specify the source address to be used by that socket. The only way I know of to specify the source address for outbound packets is to use a raw socket and compose the IP header ourselves, which would be overkill. Benny, are you aware of some other method to accomplish this? -- 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] chan_sip sending from wrong source address when multiple interfaces are used
On 07/11/2012 07:51 AM, Olle E. Johansson wrote: 10 jul 2012 kl. 20:50 skrev Kevin P. Fleming: On 07/10/2012 03:24 AM, Olle E. Johansson wrote: The Asterisk SIP channel has no knowledge about interfaces and can't bind to a specific interface for communication. In fact, it's a well known bug that if you have multiple interfaces with different IP networks, Asterisk will send from the wrong IP on some of the interfaces. Are you sure about that? The only problem area that I'm aware of is when there are multiple *overlapping* interfaces (on the same subnet, or providing the same route(s)). In that case, Asterisk can receive messages on one IP address out of the overlapping set, but reply using a different one from the set, because it doesn't specify the source IP address and instead lets the UDP/IP stack select one. If the interfaces don't overlap in any way, I don't see how it would be possible for Asterisk to send messages with the wrong source IP address, since it does not specify the source IP address at all. If this is occurring, it must involve the operating system's IP stack in some fashion. Yes, I still use quite a lot of IPtables tricks to overcome this issue. That's quite interesting; can you describe a scenario where this occurs? -- 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] Trixbox or FreePBX or Elastix or PBX In a Flash
On 07/10/2012 01:42 PM, Carlos Alvarez wrote: I'm currently trying to decide on which GUI-enabled version of Asterisk to use for one particular installation, where we will need good telecommuter support. We've made it so easy for people to work remotely that the customer is downsizing their real estate and will have 90% remote workers with them rotating through the office as needed. So most phones in the office will be shared, and I'm looking for a version of Asterisk that will easily allow people to log in and out of a specific desk. What are your suggestions? I have very little experience with GUI versions of Asterisk; we use bare Asterisk for nearly everything. This can be done using Digium phones; they have built-in support for selecting which 'user' they should be when they are reconfigured. It's slightly more complicated than a simple login/logout because it requires rebooting the phone, but it's there and it works. They even support the case where a user has a phone at their house, but comes into the office and 'steals' their extension away from the house phone; the house phone will then go into an automatic reconfiguration state and wait for someone tell it which extension it should 'be'... when the user returns home, they can 'steal' the extension back from the office. -- 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] sip.conf and binaddr issue
On 07/10/2012 03:24 AM, Olle E. Johansson wrote: The Asterisk SIP channel has no knowledge about interfaces and can't bind to a specific interface for communication. In fact, it's a well known bug that if you have multiple interfaces with different IP networks, Asterisk will send from the wrong IP on some of the interfaces. Are you sure about that? The only problem area that I'm aware of is when there are multiple *overlapping* interfaces (on the same subnet, or providing the same route(s)). In that case, Asterisk can receive messages on one IP address out of the overlapping set, but reply using a different one from the set, because it doesn't specify the source IP address and instead lets the UDP/IP stack select one. If the interfaces don't overlap in any way, I don't see how it would be possible for Asterisk to send messages with the wrong source IP address, since it does not specify the source IP address at all. If this is occurring, it must involve the operating system's IP stack in some fashion. -- 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] Trixbox or FreePBX or Elastix or PBX In a Flash
On 07/10/2012 01:50 PM, Carlos Alvarez wrote: On Tue, Jul 10, 2012 at 11:46 AM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: This can be done using Digium phones; they have built-in support for selecting which 'user' they should be when they are reconfigured. It's slightly more complicated than a simple login/logout because it requires rebooting the phone, but it's there and it works. They even support the case where a user has a phone at their house, but comes into the office and 'steals' their extension away from the house phone; the house phone will then go into an automatic reconfiguration state and wait for someone tell it which extension it should 'be'... when the user returns home, they can 'steal' the extension back from the office. Unfortunately, telling them to dump 50-some Polycom phones that still work is a tough one. Not impossible though. So if I understand correctly, the user reboots the phone when he arrives, and goes into the configure mode? Is it end user friendly, or complex? I still don't have a server with the Digium phone module on it to test. That's basically it, yes. I just did this on my D70: * Press Menu, select Restart, press Yes. * During the restart, while the progress bar is progressing, push a key. This results in 'Reconfiguration enabled' on the screen. * When the server list appears, choose one (if there is only one server, this will be skipped). * Choose a user from the user list and enter the PIN for it; alternatively, if a global PIN is required by the server, it will need to be entered before the user list will be displayed. * Wait for the phone to configure itself. This takes about 60 seconds. -- 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] Asterisk trying to call a queue with no members
On 07/06/2012 10:15 AM, Antonio Modesto wrote: Hi, I am trying to configure some static queues in asterisk, it's almost working, the problem is that asterisk is not verifying if the queue has logged members. For example, if I create queue called test, which has no members logged in, and try to place a call using Queue(test) I get into the queue, even if all phones are turned off, I tried to verify it with the QUEUE_MEMBER function, using the ready parameter, it shows me that all members are logged in. Here is my queues.conf: [general] persistentmembers = yes monitor-type = MixMonitor [default-queue](!) musicclass=default joinempty=yes This setting will cause Queue() to allow callers to join the queue even if the queue has no members. Since you are saying you don't want that, you should turn it off. leavewhenempty=no strategy=random timeout=10 retry=3 timeoutpriority=conf autofill=yes announce-position = yes ringinuse=no [recepcao](default-queue) member = SIP/100 member = SIP/101 member = SIP/102 You have statically defined the members of this queue, there is no 'logging in' required. The Queue() app does not try to determine whether the devices you list as members are currently connected to Asterisk or not. -- 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] Asterisk trying to call a queue with no members
On 07/06/2012 12:36 PM, Antonio Modesto wrote: I don't want the users to manually login in the queue, I want they join the queue when they turn on their phone. I thought that this was the right way of doing it, how can I do it? That's a reasonable way to do it if you like, although it's pretty uncommon for users to have 'turn on' their phones. -- 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] Outbound Asterisk calls default directmedia specifications
On 07/04/2012 01:47 PM, sathiish kumar wrote: Thanks for the response.. I did change it in the [general] settings.My setup is something like I have a remote conference (not meetme) which will send reinvite to redirect the RTP flow to a different server to load balance.There are three clients who join in the conference and i can listen to two other clients speak from the third client but when i record the conversation my recording of one of the clients ends before the stipulated hangup time. I am guessing this is because one of the clients doesn't understand what to do with a reinvite.. Any suggestions.In the SIP.conf i have changed the directmedia option to no and also enabled the ignoresdpversion option. The 'directmedia' option *only* controls whether Asterisk will attempt to drop itself out of the media path between two SIP endpoints. It has no effect on whether or not Asterisk will respond appropriately to a re-INVITE received *from* a SIP endpoint (to which Asterisk should always respond properly, unless the re-INVITE is malformed in some way or is unacceptable). -- 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] SendFAX timestamp
On 06/27/2012 09:30 PM, David Cunningham wrote: Would anyone else know if Asterisk allows use of SpanDSP's time zone conversion? No, SendFAX (in res_fax) doesn't currently offer the ability to do what you are asking about. -- 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] Digium IP Phones D40
On 06/22/2012 05:12 PM, bilal ghayyad wrote: One of the problems I faced with Polycom is the voice volume and ring volume, it is low. When it rings, even if it is maximum volume, still it is weak. When I talk and I set the volume to the maximum, I still feel the voice volume is low and would if to increase it. I have never, in over 7 years of using Polycom phones, heard anyone complain that the maximum volume was too low. Most devices of this type have their maximum volume controlled to meet guidelines set by government and industry recommendations (in order to avoid causing damage to users' ears), and the Digium phones are no exception. If you are finding that the volume produced by common SIP phones is too low and you can't make it loud enough, I'd bet that the problem is not in the phones, but in your environment or your ears :-) -- 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 channel 'SIP/192.168.1.69-00000000' refused to negotiate T.38
On 06/22/2012 12:05 PM, Ahmed Munir wrote: Here is my setup; Fax machine - PSTN - Cisco Voice GW - IP cloud - Asterisk. As on Cisco Voice GW, T.38 fax already configured on SIP protocol. Apparently your configuration of the 'Cisco Voice GW' was not successful, as it refused to accept a re-INVITE from Asterisk that wanted to switch the SIP channel to T.38 mode. -- 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] SIP over SSL TCP or SRTP?
On 06/22/2012 12:56 PM, Bruce B wrote: Which one of these ensures that SIP packets are sent and received in a secure format so that users using public wifi don't allow MITM type of attacks or others can't read the plaintext SIP packet info. VPN is not an option. Looking for 2nd most secure to VPN. SIP over TLS (what used to be called SSL) is what secures the SIP signaling. SRTP is for securing media streams. -- 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] TDM410 PTSN line setup with 1 analog phone
On 06/20/2012 08:44 AM, Joseph Towery wrote: Sorry to sound so much like a newb but in asterisk I am. I was initially trying to do things by hand in the extensions.conf file and had no luck. I then got from SVN checkout asterisk-gui and used it to simply try and get things started, and created a trunk, users, incoming rule, etc. from the gui and finally got dial tone, and can dial out, but I haven't got the analog phone ringing yet. I will have more targeted questions in the near future. It is just hard to find google help for analog answers. Most deal with SIP (which is my next step once I have the analog lines working). Have you read any of the O'Reilly Asterisk books? They will help you learn quite a lot about Asterisk, and they are available online. -- 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] TDM410 PTSN line setup with 1 analog phone
On 06/20/2012 09:34 AM, Joseph Towery wrote: Thanks for the tip, the answer is yes, (I forgot I copy the first message in into the body below,) but I have read a lot in the http://cdn.oreilly.com/books/9780596510480.pdf and http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html pages. I was just wanting to get the very basic analog config working prior to jumping into SIP and other higher level things, and that is where I was having a stumbling block. I am making tiny steps forward at least right now. Starting at page 79 of the 2nd Edition of the book, you'll find step-by-step instructions on setting up an FXS port for use with an analog telephone. -- 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] Clipping issue with SIP over satellite
On 06/19/2012 04:23 AM, Richard Kenner wrote: You have hardware echo canceling *outside* of your T1 card? No, on the card. Then you definitely don't want 'echocancel=no' set, or you'll disable it. The DAHDI layer has some buffering that can help with jitter, but the default buffers can only handle 80ms of jitter. You can increase this by setting the 'buffers' option in chan_dahdi.conf; each buffer is 20ms by default. I'm running 1.6.2 and it appears that this is called jitterbuffers there. Is that right? Yes. I've set it to 20 and it did indeed help quite a bit, so I tried 30. Excellent! It sounds like the lack of a proper jitter buffer (of adequate size) is the issue here, since when the audio is directed at endpoints outside of Asterisk that have them, the audio is as you'd expect it to be. Interestingly, that isn't completely true. If it goes out a SIP trunk to PSTN, it works fine, but when it goes out a SIP trunk to the SV8300 (where the T1 goes), it has the same problem. This was leading me to believe that the problem was on the 8300. Well, that doesn't disprove my statement :-) Note that I said that when the audio is directed at endpoints that have a proper jitter buffer, there is no issue. If you send the call over SIP to this 'SV8300' device and still have audio issues, that would imply that this device does not have a jitter buffer capable of handling this level of jitter. -- 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] Clipping issue with SIP over satellite
On 06/17/2012 06:43 AM, Richard Kenner wrote: Things work fine when he's talking to another Asterisk phone or to a SIP trunk provider, but when connecting to a T1, there's clipping where about 1/3 of his voice (in intervals of maybe 200ms) are removed. This sounds like an echo canceller conflict, but I've set echocancel=no in chan_dahdi.conf (I have hardware echo cancelling) and it didn't do anything. I'm forcing his codec to G729 for bandwidth reasons. The phone is an Aastra 6757iCT. You have hardware echo canceling *outside* of your T1 card? If it's an echo canceler on the card, then setting 'echocancel=no' disables it. You probably don't want to do that. The DAHDI layer has some buffering that can help with jitter, but the default buffers can only handle 80ms of jitter. You can increase this by setting the 'buffers' option in chan_dahdi.conf; each buffer is 20ms by default. As long as what are dealing with is 'simple' jitter (just delayed packets), as opposed to packet reordering, then this should help quite a bit. If you have packet reordering occurring as well, then you'll need a full-fledged adaptive jitter buffer on the channel to compensate for it. In recent releases of Asterisk, this can be done by using the JITTERBUFFER() dialplan function on the SIP channel in question, but since you didn't mention your version of Asterisk, I can't speculate whether that is available to you or not. Does anybody have any suggestions here? It sounds like the lack of a proper jitter buffer (of adequate size) is the issue here, since when the audio is directed at endpoints outside of Asterisk that have them, the audio is as you'd expect it to be. -- 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] Digium IP Phones - Teleworker Capability?
On 06/14/2012 05:23 PM, asterisk users wrote: Is there a detailed application note in the Digium wiki (or anywhere else for that matter) about these implementing features under Asterisk/Switchvox? Not yet, I don't believe. -- 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] Local Channel Resource Limit
On 06/14/2012 04:20 AM, [Digital^Dude] ® wrote: How can I set a hard limit to the number of Local channels asterisk can spawn? chan_local does not have a mechanism to do this. -- 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] Digium IP Phones - Teleworker Capability?
On 06/14/2012 04:57 PM, asterisk users wrote: We couldn't see anything about this on the Digium site, but maybe someone here can comment? Do the new Digium phones provide good teleworker functionality? Yes, I believe they do :-) The benchmark we're comparing against is the capabilities of Mitel 3300 IP systems with Mitel 5330 IP phones (running their proprietary MINET protocol), specifically: a. A Mitel phone can be easily configured for teleworker mode (select TW mode and the IP of the gateway server). The phone reboots and it is ready to be used (once the Mitel border gateway is set to recognize the unit's ID, based on its MAC address, printed on the label on the back of the phone). If the phone gets reallocated back to a directly connected office environment, a simple reset procedure brings it back. Digium phones can do something similar, and in an upcoming firmware release, there will even be features available to make this happen on a fairly automatic basis. b. You can plug in the phone virtually anywhere. It has a built-in tunnelling mechanism providing end-to-end encryption and is very tolerant of the network configuration, routers, NAT, etc. Digium phones speak SIP and RTP to the server, just like pretty much any other SIP phone. They employ many modern NAT traversal techniques and should work in most network situations. They don't currently provide encryption for signaling and media, though. c. If the link between the phone and the gateway goes down, the phone will restore itself gracefully and automatically once the network function resumes. Absolutely hassle-free to the user. I don't understand this; SIP phones don't require this at all. The phone is an intelligent device on its own. If there is no network connectivity to the server, then calls cannot be placed or received, but once connectivity is restored, operation would be back to normal. d. Users can be configured to have hot-desk functionality. The phone has a default extension assigned, but the user can be set up so that they can log in to their normal office extension number from wherever they are. Their office phone is automatically logged-out and goes to its default extension when you log in to a teleworker phone (you don't have to log out from it first). Your phone buttons, display settings, voicemail WMI and access, (everything) move to this new phone, and you can work from your home office, on the road, etc., and inbound and outbound calls work just like you were there in the office (callerid, etc). Yes, this is supported. These four features would be a big selling point for us to consider moving our organization from Mitel to Digium/Asterisk/Switchvox. How much of this can be done with Asterisk/Switchvox and, say, the Digium D70 phone with dynamic button display? Most of it, I think. Give them a try! -- 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] Deleting an inadvertent message
On 06/13/2012 07:32 AM, Nick Khamis wrote: Hello Everyone, Is there any way we can delete the following message sent to asterisk ml, instead of the actual user please? I appologize for the inconvenience however, my personal info is in the email. http://markmail.org/message/gwhg4trnw4wei74k Thanks in Advance!! No, there is not any way to do that. -- 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] Get voicemail box password from dialplan?
On 06/11/2012 10:34 AM, Chet W. Stevens wrote: I would like to be able to use the dialing extension's voicemail box password to authenticate or as a PIN code in the dialplan. Is there a best method for doing this? I could use AGI scripting but I was hoping there was a built-in dialplan means for doing this. I have used VMAuthenticate but I would like more flexibility than what this offers What do you need that VMAuthenticate does not offer? -- 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] Which Digium cards to select: AEX with EF or E or P or B?
On 06/11/2012 01:48 PM, bilal ghayyad wrote:
I need to order Digium card and not able to know which one is the best quality?
Is it that of AEX with the end E or EF or P or B?
Digium does not produce cards of differing quality levels; we strive to
have all of our cards be produced with the best quality possible.
The various suffixes you refer to indicate whether the board is a plain
board, with modules, with an echo canceller, or some combination of all
of them.
I saw those card that its slot is small (I think those that end by EF), are
they the best card?
Each variant of a particular model is the same size. All AEX410 cards
are the same size, for example, regardless of which modules are
installed or whether an echo canceller is included. Whether a card's
size makes it the 'best' card for your application is up to you to
decide, based on the system you plan to install the card into.
Really I am caring to have a card that has echo cancelation and the voice
volume is high enough (because previously I faced the problem that the voice
volume was low and I tried to resolve it without any success).
It will not matter which Digium card you choose, they will all produce
identical signal levels ('voice volume') when plugged into your
telephony circuit(s).
--
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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Re: [asterisk-users] Sangoma Card Issue SOLVED
On 06/06/2012 09:46 AM, Eric Wieling wrote: For some reason 1.4.4.x was not reading chan_dahdi.conf. When I symlinked it to zapata.conf it worked. That means Asterisk (1.4.4.x?) was built against Zaptel, not 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] BRI Installation
On 06/05/2012 12:39 AM, Klaverstyn, David C wrote: Hi Guys, All my installs are based on PRI ISDN. I now have a site that I need to install BRI. As I have not done a BRI install before I’m wanting to get some information from the people in the know if I need to do anything special. Typically I install libpri, dahdi Linux and tools, asterisk ...and then configure dahdi as one does for the required hardware. Is the same true for BRI with the exception of the libpri? I have this feeling that I need to install some other Linux drivers or something for BRI. I’ve purchased a Digium HB8 card and I don’t see any mention of this in /etc/dahdi/modules. I’ve looked over the documentation at https://www.digium.com/en/supportcenter/documentation/viewdocs/H8 but there doesn’t seem to be anything there that tells me how to configure dahdi or asterisk. If someone could give me some direction that would be greatly appreciated. The Hx8 User's Manual (here: http://docs.digium.com/H8/hx8_series_manual.pdf) has an entire chapter on software installation and configuration, including DAHDI, libpri and 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] Invite + decreasing sequence number = 500 Error?
On 05/31/2012 07:37 AM, Benoit Panizzon wrote: Hi Matt It's not a bug - decrementing the CSeq header field value is directly in violation of RFC 3261. From section 22.2: When a UAC resubmits a request with its credentials after receiving a 401 (Unauthorized) or 407 (Proxy Authentication Required) response, it MUST increment the CSeq header field value as it would normally when sending an updated request. I sent this to the developers of the C3 Softswitch. They answered by quoting this part from RFC 3261, 8.1.3.5 Processing 4xx Responses: If a 401 (Unauthorized) or 407 (Proxy Authentication Required) response is received, the UAC SHOULD follow the authorization procedures of Section 22.2 and Section 22.3 to retry the request with credentials. [...] In all of the above cases, the request is retried by creating a new request with the appropriate modifications. This new request constitutes a new transaction and SHOULD have the same value of the Call-ID, To, and From of the previous request, but the CSeq should contain a new sequence number that is one higher than the previous. Here it says it should, so a lower CSEQ is allowed and asterisk is wrong they say. Well I'll quote them the _MUST_ part of section 22.2 ... and this is why many members of the IETF community now refuse to allow SHOULD and SHOULD NOT to appear in new RFCs. They have a very clear meaning, and yet implementors choose to provide their own 'meaning'. In this case, as in all RFCs, the SHOULD here means that the implementation should choose this option, because if it does not, interoperability (or even basic operation) is likely to suffer. I'll look to see if there has been errata filed for this lowercase 'should' in RFC 3261. -- 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] Loss of RTP stream during DTMF collection
On 05/25/2012 06:30 PM, Dave George wrote: How can I enable the option to allow asterisk to maintain the RTP stream during DTMF collection? If it's the problem I hypothesized it was, you can set 'transmit_silence=yes' in your asterisk.conf file. -- 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] Dual- or Quad ISDN cards for PCI-X Slots
On 05/29/2012 01:48 AM, Michelle Konzack wrote: No, it does not fit, since PCI 2.0 is 5V and has only one notch. PCI 2.1, 2.2. and 2.3 do have two notches, because they are 3.3V. In clear, you can not insert old 5V PCI 2.0 cards into a 3.3V PCI-X slot Ahh, your real issue is voltage then, not the PCI specification that the card is compliant with. Cards can be compliant with any of the PCI versions you mentioned and still be 5V only, 3.3V only, or 5V/3.3V compatible. All modern ISDN BRI cards usable with Asterisk are both 5V and 3.3V compatible, but as you say, they aren't available in your price range. -- 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] axfer with simple CDR
On 05/29/2012 07:57 AM, Marek Cervenka wrote: is it possible with simple CDR fully describe axfer? (axfer is asterisk native, not phone function) No, it is not. CDRs (Asterisk or otherwise) are only capable of directly (simply) describing a call from party A to party B. They have no ability to describe call treatments, in-call features, or any other advanced features. Asterisk's CDRs *attempt* to represent such information, but as you've seen, they don't satisfy everyone, and it seems that many parties have conflicting ideas as to how things like transfers should be represented in CDRs. -- 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 tell VPM presence without restarting?
On 05/28/2012 05:18 AM, Tony Mountifield wrote: The wct4xxp module will log kernel messages when starting up, indicating whether a VPM module (VPM400 or VPM450) was found. I have some systems that have been running long enough that the messages files from the last reboot have long since been rotated out and deleted. They are older systems running zaptel 1.2.27. Is there any way using one of the zt tools or /proc to determine whether a VPM module is present, without doing a restart of zaptel or the machine to see the init log messages? Not with that ancient set of drivers, no :-) DAHDI (since 2.4, I think) has made this information available in /proc/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
[asterisk-users] Digium's new Community Support Manager - Rusty Newton
We'd like you all to help us welcome Rusty Newton to Digium's Asterisk development and community support team! Rusty has been with Digium for over five years, starting in the Technical Support department and then moving to a sales position where he assisted customers with Asterisk and Switchvox solutions to their business needs. Prior to joining Digium he spent more than five years in the telecom industry, installing, configuring and maintaining PBXs. A couple of weeks ago he moved into a new role (for him and for Digium), Community Support Manager. In this role he'll be the primary person responsible for ensuring that Digium's community services are providing what the community members need, that the systems are operating properly, and that issues and questions are getting the attention they deserve. He'll be working closely with our Community Director as well, especially for events like AstriCon and others. He works directly with the software development team at Digium, which will allow him to focus almost exclusively on technical issues and discussions. We're quite excited that he has taken on this role and we expect that you will soon see the benefits of his activities across the community! -- 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] Dual- or Quad ISDN cards for PCI-X Slots
On 05/25/2012 11:10 AM, Michelle Konzack wrote: I am hit by some frustrations because my Server has only two PCI-X slots and my Eicon Diva 4BRI-8M, which should work fine with Asterisk, is only PCI 2.0 standard and does not fit into the PCI-X slot. This does not make sense; PCI 2.0 cards should fit just fine into PCI-X slots. Do you mean PCI-Express instead? That's very different. -- 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] Loss of RTP stream during DTMF collection
On 05/25/2012 04:30 PM, Dave George wrote: I am using asterisk for voice mail. During DTMF collection Asterisk stop sending any RTP Packets. The gap between two consecutive packets are 4 seconds, which is huge enough to screw up the jitter buffer. When ever asterisk stops to receive DTMF, the RTP stream is cut and we loose audio. I don't have this issue when calling from a SIP phone. I only have this issue when calling from one media gateway to the asterisk box. Any suggestions welcome. Can I play some file in the back while collecting DTMF? You are missing quite a lot of crucial information required for anyone to help you. First, what version of Asterisk are you using? Second, what type of channel is being used to connect to Asterisk? You mention it works from a SIP phone, but not from a media gateway.. is that gateway also using SIP, or something else? What does 'during DTMF collection' mean? Do you mean after a prompt has been played and the voicemail application is waiting for input, or is this during prompt playback, or something else? Quite some time ago Asterisk was changed to ensure that silence would be sent while an application was running and waiting for input from the caller; if your version is older than this, then that could explain what you are seeing. That's just a mildly-educated guess though. -- 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] Detecting Fax Tones over IAX2
On 05/23/2012 08:41 PM, Cody Harris wrote: Hello All, I use IAX2 as the incoming connection from my DID provider. For whatever reason, this works best for me, SIP connections lag very frequently and only have about a 50% success rate for incoming calls (they get dropped mysteriously). I'm trying to implement a fax/voice switch. I have faxdetect=both in my sip.conf, and when I use sip, it works well. However, from what I can tell, there's no such option for IAX2 connections. Any ideas on what I can do here, or am I out of luck? It's quite hard to provide suggestions since we don't know what version of Asterisk you are using. However, in Asterisk 10, there is a channel-agnostic FAX detection function that can be applied to any channel type, so at a minimum that is one way to solve your problem. -- 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] Detecting Fax Tones over IAX2
On 05/24/2012 09:44 AM, Tim Nelson wrote: BUT, even if fax is detected on an IAX2 channel, the only reason would be to change dialplan logic accordingly correct? There is no T.38 equivalent within IAX2, which means the OP will be handling faxes over a clear VoIP channel. The information here is of utmost relevance: http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Absolutely correct. -- 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] T.38 debug logs
On 05/24/2012 09:54 AM, Arstan wrote: Dear list, I have a project where I have: Asterisk 10 --AudioCodes -- E1-- Provider AudioCodes supports T.38 and passes the faxes through E1 to the provider. From what I read, Asterisk 10 has the most stable(full) T.38 among other releases. Asterisk 10 has T.38 gateway support, but you won't be using it here because your AudioCodes device will be performing that function. Outside of gateway support, the T.38 functionality in Asterisk 1.8 and Asterisk 10 are very close to identical. My Question: Can I somehow see in the logs if T.38 packets sending and see somehow its debugs? Or I should just be better off with capturing sip data through tcpdump? This will depend on what you are asking the Asterisk 10 system to *do* with T.38. Are you sending FAXes from it, or receiving FAXes into it, or something else entirely? -- 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] twenty thousands (20, 000) users, which asterisk and how many servers?
On 05/23/2012 07:16 AM, bilal ghayyad wrote: Hi All; I need to use Asterisk for 20 000 users, so which asterisk version to be used? Is there asterisk version that supports 20,000 users on one hardware machine? Can I use one strong hardware server i7 with 64 GB RAM and fast hard desk to handle 20 000 users, and concurrent calls 2000? Or I need multiple servers, how much? If I am going to use multiple servers (until now I do not know how much, and I do not know if the barrier will be the asterisk software or the hardware), then do I have to use special SIP proxy or I have to use load balancer)? In this case, I have to use asterisk Database (so all the servers will read/write from the database)? What about AsteriskNow, can it support? AsteriskNOW is a GUI on top of Asterisk; it does not change the ability of the system to handle call load. Modern versions of Asterisk can easily handle 2,000 simultaneous calls, even with media (non-transcoded) passing through the server. We have a community member who has improved chan_sip in Asterisk 10 (and later) to be able to handle 10,000 simultaneous calls. Handling 20,000 registrations is probably more of a concern for Asterisk at this point; I've never heard of anyone attempting to handle that many on one system. In spite of all this, though, the other advice you've received in this thread is sound: even if a single system can handle the load, doing so is asking for a major problem if that system experiences a failure. You'd be much better off to at least split the load across two machines, both of which should be large enough to handle the entire load when necessary. -- 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] gr-303
On 05/21/2012 11:43 PM, Don Dawson wrote: Does asterisk support gr-303? Seems to be undocumented if so. Yes, chan_dahdi has support for connecting GR-303 channel banks to Asterisk via T1 spans. It's pretty rare that someone tries to use it, though, and as you say, there is little documentation. There is no support in chan_dahdi to make Asterisk behave *as* a GR-303 channel bank. -- 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] SET SIP_CODEC and Video issues
On 05/19/2012 03:52 PM, Tarek Sawah wrote: Thank you, Any idea how? Need to be able to control the codecs in use through soem bandwidth tests. so i need to be able to set the SIP_CODEC and still be able to do Video. any suggestions? Unfortunately you can't do what you want using SIP_CODEC; if you set that variable, the formats (both audio and video) allowed on the channel are reset to whatever you specify, and that variable can only hold one format name. It seems odd though that you want to change audio codecs based on bandwidth tests, but still allow video. The video stream is going to consume vastly more bandwidth than the audio stream. -- 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] DPMA for Digium Phones
On 05/20/2012 11:29 PM, Danny Dias wrote: I have a question regarding DPMA for Digium Phones, if i install the DPMA on my Asterisk Server A, and then, i move the phone to register into another Asterisk Server B, can i install for free another DPMA license for my digium phones on this second server? can i move the DPMA from one server to another or i have to buy new licenses? You can get as many free licenses as you like. You can also move a DPMA license from a server to another server one time; after that you need to just 'throw away' that license and get another one. There is no 'buying' licenses, they are free. -- 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] DPMA for Digium Phones
On 05/20/2012 11:48 PM, Danny Dias wrote: By the way, the DPMA is only available for Asterisk Certified, is that right? is there any problem for an Asterisk production server on a customer with Asterisk OpenSource 1.8.5 to migrate to Asterisk Certified? What is exactly an Asterisk Certified? Do we have to pay for some license? Your questions are answered on the Certified Asterisk page on the Digium website: http://www1.digium.com/en/products/asterisk/certified-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] Asterisk and the media path
On 05/21/2012 07:03 AM, David Wessell wrote: I am attempting to get an asterisk server to step out of the media path, but am running into a brick wall. Can someone assist? Here's my setup.. Ultimate SIP Provider --- LCR Trunk (Asterisk 1.6) PBX (Asterisk 1.8). In order to be able to know whether any known bugs are interfering with what you are trying to do, we need more specific version numbers. I am attempting to get the trunk to step out of the media stream. There is no NAT involved, all machines have a public IP. In the trunk's sip.conf I have: directmedia=yes directrtpsetup=yes Please turn off directrtpsetup; it's experimental and doesn't always work as you'd expect. In theory it is exactly what you want in this scenario, though. If you are using Asterisk 1.6.0.x or 1.6.1.x, 'directmedia' won't be recognized either. And on the connection to the pbx I have canreinvite=yes Why 'directmedia' on one side and 'canreinvite' on the other? They are synonyms, you should use the same name on both sides. On the pbx I have the trunk connection set to canreinvite=yes. This is unnecessary, unless the devices on the other side of the PBX are also on public IPs and you want the PBX to drop out of the media path as well. In the CLI on the LCR trunk I see: -- SIP/blahblah-000b answered SIP/1722291028-000a -- Native bridging SIP/1722291028-000a and SIP/siproutes-000b Which would make me think that the lcr trunk is stepping out of the media stream. However when I pull up a tcpdump in wireshark I still see a RTP connection? Can someone point me in the right direction? No, native bridging just means that the media stream will be bridged at the RTP layer instead of in the Asterisk core. Whether that is done using a Packet2Packet bridge in the RTP stack itself, or pushed out to the endpoints (directmedia), it's still a native bridge. However, the fact that you are seeing this message means you don't have any of the large number of reasons that would impede native bridging (transcoding, recording, etc.). It seems like you have the configuration set up (mostly) properly, so in order to know what is going on you're going to have to post a more complete log snippet, including 'sip debug' output. -- 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] Asterisk and the media path
On 05/21/2012 11:46 AM, David Wessell wrote: Hi Kevin, Thank you. Here's the requested information. 1) The Trunk is running 1.6.2.9. Also it's running a2billing. 2) The PBX is running asterisk 1.8.12.0 along with FreePBX. 3) I did directmedia on the trunk and canreinvite on the pbx since they were different versions. Sure, but you used the *old* name for the option on the system running a *newer* version of Asterisk. That's why I was confused, I suspected you might have thought that 'directmedia' and 'canreinvite' were somehow different. Since both of your systems are 1.6.2.x or later, you can use 'directmedia' on all of them. -- 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] Asterisk and the media path
On 05/21/2012 12:54 PM, David Wessell wrote: So I need directmedia set in sip.conf on the LCR trunk. 1) Do I need it in the individual trunk settings for each pbx? Or is in sip.conf enough? You say 'in sip.conf' multiple times, but that's far too vague to mean anything. sip.conf is a configuration file used to define SIP peers (and users), each with their own settings. There isn't any place you can set 'directmedia' on and have it affect all your defined peers. Each peer that should have directmedia enabled must have it set. 2) Do I need anything on the pbx side that we are hoping to transfer media to? No. 3) How long into the call before the media is transferred over? It should happen quite quickly after the call is answered. -- 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] Asterisk 1.8 canreinvite
On 05/18/2012 03:56 AM, Jonas Kellens wrote: is canreinvite still supported in Asterisk 1.8 ?? I read about directmedia being available in asterisk 1.8, but is it the same ?? What exactly did you read? 'directmedia' is the new configuration option name for 'canreinvite'; they are the *same* feature. If the document(s) you read didn't make that clear, the authors did you a disservice. -- 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] Fwd: RTP stats explaination
On 05/18/2012 12:51 PM, Steve Edwards wrote: On Fri, 18 May 2012, Dave Platt wrote: A maximum jitter of 230 milliseconds looks pretty horrendous to me. This is going to cause really serious audio stuttering on the receiving side, and/or will force the use of such a long jitter buffer by the receiver that the audio will suffer from an infuriating amount of delay. Even a local call would sound as if it's coming from overseas via a satellite-radio link. Won't a cell-to-cell call experience delays in the 300ms range? Many moons ago I remember listening with a cell while tapping on the table with another cell and being stunned with the magnitude of the delay and that most people manage to carry on conversations without noticing. Yes, cellular networks have largish latencies, but no jitter. -- 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] realtime configuration for /etc/dahdi/system.conf
On 05/17/2012 07:38 AM, Kamlesh Kumar wrote: can we load the settings of /etc/dahdi/system.conf from database table in real time. No. Since these settings are associated with physical connections on the server, it seems quite unlikely that being able to change them in realtime would ever be useful. -- 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] R-Series with NON-DIGIUM card on servers
On 05/12/2012 12:07 PM, Danny Dias wrote: What about the Database and recording calls replication? as i could see, the RSeries does not take into account these data. The Digium R-series devices are electronic switches used for routing telephony circuits; they don't have any part in the actual failover process, data replication, or anything of the sort. All of those functions need to be handled via software on the server(s) involved. The R-series user's manual describes one way this can be done using Asterisk and open source tools commonly available on Linux distributions. -- 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] R-Series with NON-DIGIUM card on servers
On 05/11/2012 10:46 PM, Danny Dias wrote: Hi, I would like to know if the servers (A and B) could use boards non-digium with the R-Series HA product from Digium, i have a couple of B600E Sangoma to put on each server and use the R-series to provide HA. Is that possible? Yes. The Digium R-series failover appliances will work with any device that uses the appropriate type of PSTN circuits (digital or analog, depending on the R-series model), even a legacy PBX. -- 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] Digium IP Phones
On 05/11/2012 11:08 PM, Danny Dias wrote: Does the D40 will support the option to develope apps? As i could see on videos only the D70 has the apps button, and also, the lcd screen is smaller. Right? All of the Digium IP phones will support user-developed applications once the SDK has been released. The D40 and D50 have the same size screen, but it is smaller than the one on the D70. The D40 and D50 do not have a hard 'Apps' button, but they do have an on-screen softkey for access to applications. -- 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] Least Machine Specs to run a production asterisk server
On 05/11/2012 10:13 AM, eherr wrote: Well.. I would say that there would be about 40 Polycom Soundpoint IP335s G711u I would like to restrict this to say 15 inbound lines and 40 outbound lines. Probably looking at 8 concurrent calls. This doesn't add up; you say '15 inbound lines' and '40 outbound lines', but 8 concurrent calls. What are these 'lines' going to consist of, and what will they be used for when they aren't being used for calls? The amount of load you are talking about could be handled by a small Soekris box, a router running OpenWRT, a SheevaPlug, or any number of tiny, low power embedded devices. As Carlos already said, *any* decent x86 box produced in the last five years would be able to handle this without any noticeable CPU load at all. -- 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] Replacing PBX with Asterisk, need feedback on my new architecture.
On 05/06/2012 01:39 PM, Paul Belanger wrote: 800 SIP phones on one server? I wouldn't want to do it. Add a SIP proxy to your design and have it handle all your SIP. Then you can load balance across multiple asterisk boxes. You'll be thankful you did this at the start, as it will allow you to increase resources more easily. As has already been pointed out by others in this thread, 800 phones on a single Asterisk server (using Asterisk 1.8.x or later and a decent spec server) is really no problem. If all of those phones are going to be subscribing to hints for a dozen or more of the other phones, then yes, that could be an issue, as the amount of NOTIFY traffic would be quite high... but for registration and normal calling, even if all these phones were in use at once, I would not expect any issues at all due to performance. The other comments about being able to take down a server for maintenance and not lose calling ability are certainly worth taking into consideration as well, but if your planned deployment would allow for reasonable scheduled maintenance windows, even that wouldn't justify the complexity of adding in one SIP proxy (or a pair of them) to the equation. -- 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] Replacing PBX with Asterisk, need feedback on my new architecture.
On 05/11/2012 04:53 PM, Alex Balashov wrote: Are you certain that this wouldn't be an issue if the phones had low re-registration intervals? Historically, I've seen the Asterisk registrar faceplant with throughput in excess of 5-7 registrations/sec, though I have no idea as to whether that holds true of newer releases. Yes, frequent registrations could be an issue, but when deploying a new systems it would be wise to not configure the endpoints that fashion :-) The reasons that people used to do so can now be addressed via other mechanisms that don't carry the performance penalty of registrations. However, it would not surprise me in the least if Asterisk 1.8.x and later handled that volume of registrations without much of a problem. -- 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] Digium IP Phones
On 05/09/2012 08:38 PM, Danny Dias wrote: Hello, Im looking to buy a digium phone D70 unit just for testing on lab; to really understand the phone and features. I cant find any website with opinions; any here? Are they really valuable to the price? (D70 quite expensive) Does the SDK for building apps is usable? Can you build powerfull apps? Examples? The phone app SDK has not been released yet, it's still under development. -- 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] where can i find code documentation
On 05/10/2012 05:08 AM, Arif Hossain wrote: Its rather surprising that i'm unable to find the code documentation generated by make progdocs. It should be /usr/share or /usr/local/share but it does not appear to be there. Any clue? It is generated in the 'doc' directory of the source code tree. -- 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] enabling dialing by sip uri
On 05/10/2012 09:39 AM, Arif Hossain wrote:
I have following sip account :
Name/username HostDyn
Forcerport ACL Port Status Description
demo-alice/demo-alice 192.168.7.47 D
N 1080 Unmonitored
demo-bob/demo-bob 192.168.7.47 D
N 5060 Unmonitored
and i have set up the following extensions for them:
ASTERISK_IP=192.168.7.39
[users]
exten=6001,1,Dial(SIP/demo-alice,20)
exten=6002,1,Dial(SIP/demo-bob,20)
exten = _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}]?unhandled)
exten = _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}:5060]?unhandled)
exten = _.,n,Macro(uri-dial,${EXTEN}@${SIPDOMAIN})
exten = _.,n,HangUp()u
[macro-uri-dial]
exten=s,n,NoOp(Calling as SIP address: ${ARG1})
exten=s,n,Dial(SIP/${ARG1},60)
But if i dial sip uri the call does not happen. asterisk cli shows
extension is rejected.
Asterisk is not a SIP proxy. If you are entering a SIP URI into your
phone, and that URI does not resolve to the Asterisk server as its
target, then the INVITE request sent by the phone should not even be
sent to Asterisk at all (it should go to wherever the URI resolves to).
--
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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Re: [asterisk-users] looking for solid state like PC suitable for Asterisk
On 05/10/2012 03:49 AM, Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Just a small comment here... I really find it quite humorous that people use 'solid state' to mean 'no moving parts'. All of the parts of my computers that move are still composed of solid materials, and the electrical currents involved in them still move through solid materials :-) -- 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] British Telecom ISDN BRI line issues
On 05/10/2012 03:20 PM, khalid touati wrote: Thank you Patrick for the detailed info, it does make perfect sense to me, I never expected that Digium cards have such an problem! There are patches in the works already (being tested by users in Europe) to deal with this layer 1 issue. Upcoming releases of DAHDI and Asterisk should have support for it. -- 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] enabling dialing by sip uri
On 05/10/2012 03:36 PM, Arif Hossain wrote: Asterisk is not a SIP proxy. If you are entering a SIP URI into your phone, and that URI does not resolve to the Asterisk server as its target, then the INVITE request sent by the phone should not even be sent to Asterisk at all (it should go to wherever the URI resolves to). I'm using the asterisk's ip to form sip uri at the sip client. So it resolves to asterisk no doubt. You'll have to provide more details (primarily a CLI log) then in order for anyone to be able to help you. You said that Asterisk shows extension is rejected, but extensions don't get rejected. Extensions can be 'not found', but that's very different from rejected. -- 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] Belgian BRI (euroisdn): what to use for a B410P
On 05/09/2012 12:59 PM, Bart Coninckx wrote: Hi, I'm experiencing difficulties to get a B410P running with Asterisk 10.3.1 and DAHDI 2.6.1. Am I supposed to use DAHDI for this card and ISDN BRI for my country (Belgium)? That is the supported method to use in standard Asterisk, yes. DAHDI, libpri and Asterisk working together. -- 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] Asterisk 1.8 Transfer CallerID
On 05/08/2012 08:50 AM, Jonathan Rose wrote: - Original Message - From: Jonas Kellensjonas.kell...@telenet.be To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Sent: Tuesday, May 8, 2012 7:13:30 AM Subject: [asterisk-users] Asterisk 1.8 Transfer CallerID Hello, when a call comes in and is answered by colleague A, this colleague A sees the CallerID of the external calling number. When colleague A transfers the call to colleague B, attended or unattended, then colleague B sees the number of colleague A on his screen while talking to the external calling number. That would be because this is the expected behavior. The call isn't coming from the outside caller, it's coming from the person who transferred it. I expect here that colleague B would see the external calling number on the screen of his IP-phone. How can I get this behaviour ? Thanks. Jonas. Getting this behavior shouldn't be too hard I wouldn't think. First, be aware that the Dial command has an option s(x) which is described: s(x): Force the outgoing callerid tag parameter to be set to the stringx. Works with the f option. So if you simply transfer to a dial application with that option, you can force the callerid to be whatever you want it to be. You can also retrieve the callerid of the original caller and put it on your transferring peer in a variable when starting the call. I'm not exactly sure on the specifics of that right now, but I'm pretty sure it should be possible. So then when you are making the transfer to dial, you just use that variable as your argument to the s option. This is overkill, although it is certainly a way to approach it. If the OP's SIP peers for his phones are configured to send Remote-Party-ID or P-Asserted-Identity to those phones, then the behavior he's looking for should be automatic, since Asterisk 1.8.x has Connected Line update support. Of course, this also assumes that the phones in question have the ability to receive this information and update their displays. This can be tested by using the CONNECTEDLINE() dialplan function to send anything desired to a phone that is in a call with 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] TDM400P: Lifetime Replacement
On 05/06/2012 11:42 AM, Greg Woods wrote: Second, since the parts of this card are very expensive, I am wondering if these symptoms likely mean that the main board of the card is dead, but the FXS and FXO modules might still be good. In that case, I could just get a new main card and move the modules to the new main card. The problem is that I can't find any TDM400P cards anywhere, all I can find are TDM410P's. Will the modules I have (assuming they are still good) work with a TDM410P? Yes, the module are compatible with a TDM410P, or any other Digium card supporting analog modules except the TDM2400P. -- 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] End-To-End Secured Communications
On 05/03/2012 07:17 AM, Fernando Berretta wrote: Hi, I'm analyzing how to make Asterisk communications secured End-To-End, and not sure which is the best approach, SRTP + TLS seems to be secured but.. at least by default, doesn't appear to be End-To-End allowing Asterisk administrators to wiretap communications.. some sites I've hear that with SRTP is also possible End Points exchange keys between them directly avoiding Man in the Middle, is it possible with asterisk ? how On the other hand I've found ZRTP seems to be secured end-to-end, but we couldn't find any IP phones with support for it.. just SoftPhones Could someone please point me to the right direction ? This is a fundamental architectural issue with all back-to-back User Agents used in SIP networks. They are pretty much by definition a 'man in the middle'. If they are used, the administrators will have access to call signaling and media for all calls passing through them. It is also important to realize that if you want end-to-end media security, then you would not be able to use any of Asterisk's features that involve media handling (transcoding, recording, whispering/spying, music-on-hold, conferencing, etc.) Given that, what you really want is a pure SIP proxy like Kamailio or OpenSIPs. -- 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] Playing Multiple File ( simultaneously ) on Channel
On 04/27/2012 09:01 AM, shayne.al...@gmail.com wrote: I am interested to know if there is any application or way to help me for this Scenario: When we put Callers in Q, the MOH will stop for announcements! how if we able to increase the MOH (RT/TX) and then play any announce with greater RX/TX ( and there so louder ) on the channel! without stoping MOH? This is an interesting idea, but at this time Asterisk's app_queue has no ability to do what you are asking for. -- 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] Playing Multiple File ( simultaneously ) on Channel
On 04/27/2012 10:57 AM, shayne.al...@gmail.com wrote: Yep, But I think that is can be done, independent to Queue app. If we have an application which call MusicOnHold inside! maybe we can control simultaneously playing back of files... if we have such ability then it can be replaced by what exist inside the Queue. or maybe an application which written from scratch... can be help full. It's hard to parse what you are saying, but yes... it would be possible for someone to write code to do what you want to do. -- 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] Hangup Cause and SIP Response Code
On 04/25/2012 05:29 PM, Eric Wieling wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Wednesday, April 25, 2012 6:25 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hangup Cause and SIP Response Code On 04/25/2012 04:45 PM, brya...@zktech.com wrote: Kevin I am using 1.8.x 10.x Then you have SIP_CAUSE available, although you'll have to enable it because it is off by default due to performance concerns. Does anyone know what kind of performance hit you take from SIP_CAUSE when you are using few or no calls using chan_local? The performance impact will be directly related to the number of outbound SIP channels you create; no other channels will be involved. We had a Digium OEM customer observe a 50% call load capability decrease when they started using SIP_CAUSE, but that was on a pretty busy system, and all the channels were SIP channels. -- 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] Master Registrations?
On 04/27/2012 02:28 PM, Bryant Zimmerman wrote: I am expanding our systems and am looking for a way to track wich asterisk server a peer is registred on. Here is an example. 4 Asterisk servers. An peer could regester on any of the three servers. When I a peer registers I need to some how trigger an update to a master database to tell wich server that peer is on. Then when a call comes in I can look up in the master database where that peer is an and route the call to the correct registration servers. How can I trigger a script on peer registation to update my master database table? I am under the impression that I do not want to share the same realtime peer table accross all the servers. so how would I best pull this one off? This has been done many times before, using DUNDi; it's one of the primary things that DUNDi was designed for. There is no central database to update, instead the target device is located when a call requests it. There are many examples of this on the Internet... use your favorite search engine. -- 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] CONNECTEDLINE() updated during SIP events?
On 04/25/2012 11:54 AM, Steve Davies wrote: A further question... It appears that for SIP endpoints, this facility only updates RPID and PAI headers? I have found that there appear to be 4 different SIP CID-update mechanisms out there as follows: - Update RPID and PAI (ITSP and trunks often understand this) - Update Contact: header (Aastra handsets use this) - A SIP INFO packet if Supported: callerid is specified (Older snom firmware uses this) - Update From: header if Supported: from-change is specified (RFC 4916, snom, Yealink) Are there existing plans to support any of these other methods? If not, I will almost certainly add them for my own use, and submit the code. No, we have no plans at this time to go beyond RPID and PAI support. Those two appear to cover all the current endpoints that we have been able to test with, and many community members have also used with other endpoints and had success. Changing the Contact header seems quite wrong; the display-name in a URI in the Contact header is pretty much irrelevant. Changing the From header also seems wrong; that should indicate who sent the initial INVITE, not who redirected it. I don't think we'd want to merge patches that added support for either of those mechanisms. -- 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] Hangup Cause and SIP Response Code
On 04/25/2012 07:08 AM, Bryant Zimmerman wrote:
I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
track the actual SIP response code as well. How do I get access to it
durring the hangup?
It's rather hard to answer that question without at least knowing what
version of Asterisk you are using. In some versions there is a SIP_CAUSE
feature that can be used to extract that information (although this has
been reimplemented for Asterisk 11 in a way that doesn't affect
performance as much as the old method did).
--
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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Re: [asterisk-users] Hangup Cause and SIP Response Code
On 04/25/2012 04:45 PM, brya...@zktech.com wrote:
Kevin
I am using 1.8.x 10.x
Then you have SIP_CAUSE available, although you'll have to enable it
because it is off by default due to performance concerns.
Bryant Zimmerman (ZK Tech Inc./interNetGR)
(616) 855-1030 Ext. 2003
On Apr 25, 2012, at 5:00 PM, Kevin P. Flemingkpflem...@digium.com wrote:
On 04/25/2012 07:08 AM, Bryant Zimmerman wrote:
I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
track the actual SIP response code as well. How do I get access to it
durring the hangup?
It's rather hard to answer that question without at least knowing what version
of Asterisk you are using. In some versions there is a SIP_CAUSE feature that
can be used to extract that information (although this has been reimplemented
for Asterisk 11 in a way that doesn't affect performance as much as the old
method did).
--
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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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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Re: [asterisk-users] asterisk 1.4.39 and dahdi 2.6
On 04/21/2012 04:07 PM, bilal ghayyad wrote: Dear; The output of the ./configure that is related to dahdi is: checking for DAHDI_RESET_COUNTERS in dahdi/user.h... yes checking dahdi/tonezone.h usability... yes checking dahdi/tonezone.h presence... yes checking for dahdi/tonezone.h... yes And the dependecies of the chan_dahdi as I saw in the make menuselect is (but really I do not know what the M and E means, and how I can be sure if they are existed or not)? Depends on: res_smdi(M), dahdi(E), tonezone(E), res_features(M), pri(E) Well, unfortunately 'menuselect' doesn't actually tell you what is missing. We'll need to see the entire output from the 'configure' script to be able to tell you what is missing. By the way, why are you trying to use Asterisk 1.4.39 and not 1.4.44? Even if you're going to use an unsupported branch, you should still use the most recent release from that branch. One final note: please don't reply to list posters' personal email addresses unless they ask you to do so. The list is configured to force replies to go back to the list, and that's done for a reason. -- 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] asterisk 1.4.39 and dahdi 2.6 on Ubuntu
On 04/19/2012 05:59 PM, bilal ghayyad wrote: Dears; I see this at the /var/log/asterisk/messages: [Apr 20 01:49:48] ERROR[1657] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory If you aren't using a DAHDI transcoding card, then you don't need to load the codec_dahdi module in Asterisk. Since it was built, though, you clearly have DAHDI built and installed properly, and the Asterisk build process was aware of that. Again, I am installing asterisk and dahdi at Ubuntu (uname -a Linux House 3.0.0-17-server #30-Ubuntu SMP Thu Mar 8 22:15:30 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux). I do not know if you were talking about the messages logs or about someting else? Anyway, these are the logs that I see at the messages after running /etc/init.d/asterisk restart: [Apr 20 01:49:48] NOTICE[1657] cdr.c: CDR simple logging enabled. [Apr 20 01:49:48] NOTICE[1657] loader.c: 142 modules will be loaded. [Apr 20 01:49:48] WARNING[1657] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. [Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: Starting AEL load process. [Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. [Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. [Apr 20 01:49:48] ERROR[1657] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory All of that is perfectly normal. If you want that ERROR message to go away, add 'noload = codec_dahdi' to your /etc/asterisk/modules.conf file. -- 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] asterisk 1.4.39 and dahdi 2.6
On 04/20/2012 03:17 PM, bilal ghayyad wrote: Dear; Well, I did make menuselect and I really found the XXX and did not get the ability to select the channel. So what could be the reason? When you are in menuselect, looking at the 'channels' page, scroll the cursor down to chan_dahdi (marked with 'XXX'), and look at the bottom of the window/screen. In that area there will be information about the chan_dahdi dependencies that were or were not found by the the configure script. If you can copy and paste that information here, we can try to help you figure out what is going on. It's quite strange that codec_dahdi successfully built but chan_dahdi did not; the problem is likely not related to DAHDI, but due to some other dependency that chan_dahdi has. As I said before, what we really should be looking at is the configure script output that indicates what it was able to find and what it was not able to find, but the menuselect information is a reasonable next step. -- 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] asterisk 1.4.39 and dahdi 2.6
On 04/18/2012 06:43 PM, bilal ghayyad wrote: Dear Warren; Yes I am compiling and installing dahdi first and then I start by asterisk 1.4.39 but I do not find chan_dahdi under /usr/lib/asterisk/modules, but if I used asterisk 1.8, it is working fine. From the other side: I tried asterisk 1.4.44 and same thing (I am not able to see the chan_dahdi) !! By the way, I am using ubuntu. Which asterisk 1.4 version that you tried it with dahdi and you were able to find the chan_dahdi? Really I tried too many attempts and until now I am not able to find a solution ! What I am missing? You are asking people to help you guess what is wrong, instead of looking at the output of the Asterisk configure script. When the configure script checks for DAHDI, if that process fails for some reason, it will tell you why. That information is required for anyone to be able to help you. -- 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] g729 freezes 1.8
On 04/19/2012 10:57 AM, samuel wrote: Just in case it helps: It turned out that from asterisk version 1.8.4 on, the g729 binaries are different from the previous versions so it was a version mismatch between the g729 (1.8.0_3.1.5) and asterisk (1.8.8 and higher). Ahh, and that is 'documented' on the G.729 download selector page here: http://www.digium.com/en/docs/G729/g729-download.php Did you use that download selector, or go directly to the downloads.digium.com site to grab the files? -- 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] g729 freezes 1.8
On 04/19/2012 11:52 AM, A J Stiles wrote: On Thursday 19 April 2012, samuel wrote: Just in case it helps: It turned out that from asterisk version 1.8.4 on, the g729 binaries are different from the previous versions so it was a version mismatch between the g729 (1.8.0_3.1.5) and asterisk (1.8.8 and higher). Thanks to the Digium support department that found out the issue. Someone really needs to get the mPlayer folks (based on the Continent, where mathematics is not patentable) to create an Open Source g729 codec implementation . Source code availability is not the issue; the reference source code is easily obtained from the ITU-T. Many of the G.729 patent holders are companies based in Europe, so I suspect they would have a different opinion than you do about the legitimacy of their patent claims on G.729 :-) In any case (and of course IANAL), it is my understanding that the patents that cover the base G.729 recommendation, along with Appendices A and B, will all expire in the next year or so. We'll have to see what that means for the market, especially with new, more freely licensed, codecs coming out that provide substantially better performance. -- 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] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
On 04/18/2012 06:08 AM, Niccolò Belli wrote: Hi, Il 18/04/2012 00:39, Kevin P. Fleming ha scritto: You guys know that it works in Asterisk 10, but you say you can't use Asterisk 10 for some reason that I don't understand. 1) No Debian packages for v10. If you have to maintain lots of servers, installing from sources is a big burden. Compile, install and forget isn't the way I work: if I have to apply a fix or close a security hole I can easily push the patches to my build server which will recompile all the branches I maintain, then every server will automatically upgrade with cron jobs. This is a valid point, and we'll get this corrected. Our package repository should have packages for Asterisk 10, but it doesn't. 2) A new whole of problems when upgrading production machines from a working 1.8.x to v10. That will mean parsing configs manually, find the problems and fixing them. I haven't seen any rash of problems with config files when users upgrade from 1.8 to 10; in fact, we've changed development policies specifically in order to avoid breaking existing working configurations during upgrades, except when they are unavoidable. 3) Third parties utilities/hardware/modules. I'm still waiting for a fix for my Sangoma BRI card which did broke when upgrading... You need a compatible version of third parties components to use recent versions of asterisk/dahdi/whatever and upgrading third parties components does always mean problems. Do you expect Debian-style packages to include these third-party components in Asterisk? If you are talking about DAHDI specifically, moving to Asterisk 10 does not change DAHDI requirements at all. 4) Isn't v10 supposed to be beta/non-production/non-long-term-support?[1] If we want to honor what Digium says we should use 1.8 for production servers when reliability is important. Backporting a single unstable feature is much better than the whole thing. Asterisk 10 is not 'beta' or 'non-production', I have no idea where you are getting such an idea. Yes, it is a 'standard', not 'long term support' release, but it is still fully supported and intended for production use (it is not a 'developer' release). If you want Digium to be able to support your installation, especially for a long term, adding in a series of complex patches that significantly change behavior will not lead to a supportable system; if you report an issue against your patched version of Asterisk, the first response will be to replicate the problem without the patches in place, which defeats the purpose of using a 'supported' release. 5) What was the purpose of the t38gateway-1.8 branch? Why did it existed at all if not to allow users to use t38 gw in production servers? I even read about the possibility to backport t38 gw to 1.8 as a plugin, but it seems it isn't a requested feature (which is strange because I know peoples who stopped using asterisk because of the lack of t38 gw). You'd have to ask the community developer who created the branch what his intentions were with it; it's not an 'official' release of Asterisk, and at this point it isn't supported by anyone. The T.38 gateway code was significantly reworked to get it merged into trunk (which became Asterisk 10), because the 1.8 version had a lot of serious issues. That code is most definitely *not* ready for production, especially given how difficult T.38 interoperability is in general. T.38 gateway support isn't available as a 'plugin' for older releases because those releases don't have the necessary APIs and functionality needed to make it work. Adding those into an older release would risk destabilizing that release, and would dramatically increase the testing and support burden. I really don't want to do polemics: I always used pstn for the faxes until now and I will keep using it. No problem. If you feel that having a discussion about what makes sense for users to do and not to do is 'polemics', then fine, you can do whatever you like. Just please stop trying to assign blame or fault to people because this old, unsupported branch doesn't do what you want, especially when there is a current, fully supported release that will do what you want. -- 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] g729 freezes 1.8
On 04/18/2012 06:13 AM, A J Stiles wrote: On Wednesday 18 April 2012, samuel wrote: On 18 April 2012 10:33, A J Stilesasterisk_l...@earthshod.co.uk wrote: Are you sure your g729 module, your Asterisk and your kernel are of the same bittedness? I'm pretty sure it's not a problem of 32-64 bits: Asterisk 1.8.11.0 built by root on a x86_64 running Linux on 2012-04-18 07:45:43 UTC and I downladed the binaries from http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.8.0/x86-64/ And asterisk loads the module, as you can see in the log files I sent. So it doesn't look like a problem with 32-64 bits Ah, well. It's always worth a shot, though. It could still be a missing library; run `ldd` on the .so file(s), and make sure all needed libraries are installed. The simplest route to solving this problem is to contact Digium's support department; this is a Digium commercial product and you are entitled to technical support. The simple answer to your question is no, there are no known incompatibilities between Asterisk 1.8 and Digium's G.729 codec modules (if there were, we'd fix them). -- 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] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
On 04/18/2012 11:23 AM, Dan Austin wrote: Kevin P. Fleming wrote: This is a valid point, and we'll get this corrected. Our package repository should have packages for Asterisk 10, but it doesn't. How likely is it that a Centos 6 repo might be setup at the same time? It's on our list, but since the RPMs are primarily designed to support AsteriskNOW, and AsteriskNOW is still built on CentOS 5, it's not a high priority. -- 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] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
On 04/17/2012 06:17 AM, Larry Moore wrote: The send log you have posted does not show any outgoing T.38 packets from your system. I set up a test build of 1.8.11.0 using the patch recently released, I have difficulties sending T.38 with this patch, in fact I cannot send successfully however I can receive. I did however observe some outgoing T.38 packets. The analogue fax modem I was dialling into is under my control hence the log files showed there was no signalling coming from my ITSP, The T.38 session on my Asterisk server show the CRP's which were sent from the analogue fax device during the negotiation. The patch I have used for a while seems to give me outgoing functionality as well as incoming. I can't reproduce your scenario whereby your T.38 session is communicating with a different gateway to the SIP server you use hence can only speculate that Asterisk has difficulty wanting to send T.38 SDP traffic when it is a different device than the SIP server it negotiates with. We know for a fact that Asterisk has no trouble with the signaling and media going to different addresses/ports. Honestly, I just don't understand why all of this effort is being put into trying to use an old (and clearly broken) patch for adding T.38 gateway support to Asterisk 1.8. You guys know that it works in Asterisk 10, but you say you can't use Asterisk 10 for some reason that I don't understand. -- 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] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)
On 04/14/2012 07:33 AM, Niccolò Belli wrote: Il 04/04/2012 07:45, Anton Kvashenkin ha scritto: Check it out, thank you. You're welcome. New packages against dahdi-linux-2.6.0, dahdi-tools-2.6.0, libpri 1.4.12+svn20120409 and spandsp-0.0.6~pre20: http://www.linuxsystems.it/2012/04/asterisk-1-8-11-0-debian-squeeze-packages-with-t-38-gateway-queue-hints-and-fixed-rfc4235/ If someone can help me there is a bug with T38 gw and eutelia: http://lists.digium.com/pipermail/asterisk-dev/2012-April/054681.html Asterisk 10.4-rc1 does work, so it should be a matter of identifying the problem and backporting the fix. Keep in mind that the T.38 gateway code was reworked rather substantially when it was merged into Asterisk 10; the last version that irroot published for Asterisk 1.8 was long before this rework occurred. It's quite unlikely that locating a simple difference will actually occur, or that it would be easy to backport. -- 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] T.30 Fax and Echo Cancelation
On 04/13/2012 02:25 AM, Anita Hall wrote: I know that all echo cancelers are supposed to shut down when they hear the CNG or CED fax tone. Is this always a good thing ? I know that T.30 is supposed to be half duplex, so echo should not be there but in my session recordings, I have seen both sides trying to send at the same time (is that a sign of some other bad thing?) Your statement is not quite correct. G.168-compliant echo cancellers don't respond to the CNG tone at all, and their response to the CED tone varies. Some will completely disable themselves, others (modern and better ones) will continue to operate, but without their NLPs (non-linear processors) engaged. All FAX modems are half-duplex, but that doesn't mean that echo won't occur. Echo occurs regardless of whether the modems are half- or full-duplex. If you see both FAX endpoints sending modem data at the same time, then yes, you have some other type of problem. -- 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
