Re: [asterisk-users] fax / t38 gateway
On Oct 24, 2008, at 12:49 PM, Wilton Helm wrote: I've been following this thread and trying to sort out what is wanted, what is available, and why. Comments to the following would be appreciated and might be useful to others. 1. Why would anyone originate a FAX via VoIP? If it has to go through a bunch of translation steps at both ends, it would seem better to simply scan the document (assuming it isn't in electronic form to begin with) and attach it to an E-Mail. Because I don't want to pay to have one single POTS line running into my office when it's sole purpose is for fax. I would much rather get that fax into my asterisk in some intelligent manner and have one less company to pay. 2. Why would anyone terminate a FAX call coming through Asterisk in a FAX machine? Isn't there a way to capture it electronically? If so, it seems that putting the electronic documents in a queue where people can open them, save them, and if they wish, print them would be much more useful (and planet friendly, since a lot aren't worth putting on paper). I fully intend to do this, once the faxes gets into my * I email it off to someone. IMHO, there are only three realistic needs: A. Electronic end to end document transfer which is best done with E-Mail and not telephony. As others have already mentioned, faxing is still around due to user ease and legal reasons. : ( B. Receipt of FAX from outside (old school) sources, which is best done electronically. I agree. C. Generation of FAX to outside (old school) destination, which could be done either electronically or in the traditional manner. My user base is fairly intelligent, so I will most likely be doing this, but we still have a need to be able to fax someone outside our organization, for that reason we still must have a reliable way to get faxes to plain old fax machines. I hope to do this with a solution that converts an email into a fax then sends it off into the normal analog tubes. If end to end FAX is desired, is there any reason why Asterisk should treat it any differently than any other call? The FAX machines on each end generate and decode the information, VoIP is simply an audio channel through which is passes. Voip is a bit too loose of a term for describing this process when it comes to faxes. For the audio signals your speaking of to be recognized the standard analog audio range must come through clearly, lossy voice codecs hack that spectrum up to save bandwidth. While this is ok for humans, it kills the fax signal. This is what t.37 and t.38 try to fix. You can go read up on them elsewhere, but basically they allow faxes to be sent in a packetized manner that can coexist well on a data network. The trouble obviously being that normal (most anyways, some support t.38... as Steve`Underwood has been saying, albeit somewhat buggily) fax machines can't understand this. This is why t.38 must be decoded at some point before getting sent to a plain fax machine. And that is why I am looking for a way to decode/encode my faxes with t.38, to avoid having to pay a phone company for one analog line that would get used at most a few times a week. I don't know what T38 defines or implies, but if it is anything other than how to electronically decode a voice call that happens to contain FAX information (rather than passing it on to a real FAX machine) then I'm not sure what use it is. It would seem to me that the OP needs a way to electronically capture calls that turn out to be FAXes. Right, see above. Wilton Anyways, I hope that helped you understand my desire to set things up how I am trying to. I'm sure a lot of people are trying to do this / have tried / have done it. Hopefully this thread will help those having trouble figuring out how this is done when they start googling. : ) Brendan Martens___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hammering imap vmail storage
I found this in the sample voicemail.conf: ;pollmailboxes=no; If mailboxes are changed anywhere outside of app_voicemail, ;; then this option must be enabled for MWI to work. This ;; enables polling mailboxes for changes. Normally, it will ;; expect that changes are only made when someone called in ;; to one of the voicemail applications. ;; Examples of situations that would require this option are ;; web interfaces to voicemail or an email client in the case ;; of using IMAP storage. ; ;pollfreq=30 ; If the pollmailboxes option is enabled, this option ;; sets the polling frequency. The default is once every ;; 30 seconds. Maybe that will help? Brendan Martens On Oct 26, 2008, at 12:24 AM, Brian J. Murrell wrote: I've configured asterisk 1.4 to use imap storage for voice-mail and while I'm happy with it generally speaking it really seem to hammer the IMAP server. It appear, from the IMAP server logs that it's polling the imap server every *second* for mailbox updates for the users' voice-mail folders. Is it really necessary to do this once a second? Is this tunable anywhere? Thanx, b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax / t38 gateway
Quite right... And so we can all stop repeating ourselves; Steven has already done a great writeup on all this: http://www.soft-switch.org/foip.html Brendan Martens On Oct 27, 2008, at 3:20 PM, Kristian Kielhofner wrote: On Mon, Oct 27, 2008 at 2:49 PM, Wilton Helm [EMAIL PROTECTED] wrote: Thanks Brendan for the explanation. There is one other idea that struck me, but again, I don't know if it has any merit. My thinking is to keep FAX as FAX and electronic as electronic, rather than introducing a new hybrid approach. Obviously Entering FAX from an electronic source is as old as the FAX modem, and Exiting it electronically is as old as E-FAX, not to mention other alternatives. Is it feasible to simply specify the codec as ulaw or alaw (depending on jurisdiction, I forgot the g numbers) for calls originating from the FXS or whatever the FAX is coming from? Obviously, the bandwidth would be higher in that case, but you can't get around the laws of physics. Yes it is lossy compression, still, but it is the simple, predictable form of lossy compression that the modem in every FAX machine already is programmed to cope with. The only problems I can see would be if the provider who handles the call refuses to accept that codec, or transcodes it to something else. I don't know the likelihood of either of these. Wilton Wilton, Many providers will allow you to do faxing via g711u/g711a (G711u mu-law is used in T countries, G711a a-law is used in E countries). Of course they will allow it - fax modems talk to each other just like we do. They're just doing it with much less tolerance to error and variations in the audio. The provider's gateways, however, should detect the fax tone and disable echo cancellation, etc. What this discussion is forgetting are the issues inherent with packet networks: - latency - jitter - packet loss Standard fax machines communicating via some ATA with a G711u RTP stream cannot correct for these situations. In some severe cases. the modems might not even be able to train. V.x modem standards were not designed for packet networks. For this reason many faxes (especially at higher speeds) will fail (depending on the state of the network) when using a G711*, pass-through, or clear channel codec. You will have a much higher rate of success faxing with G711u over your LAN than a congested cable modem, for instance. That's what T.38 is for. It doesn't even use RTP, it uses UDPTL (UDP Transport Layer) or TCP (rare) to manage the transport of data and correct for transmission errors in various parts of the OSI stack. As we've said before the support for this standard varies and often times just doesn't work. - G711u will fail depending on the condition of the network. - T.38 will fail depending on the type(s) of equipment used. Faxing via VoIP is largely a crap shoot. However, it is important to focus on T.38 because I feel these interop issues can *eventually* be resolved. No one is ever going to fix the issues with packet networks*. That's why they are packet networks. We will have much better luck working towards T.38 interop. * Obviously they are some fixes like MPLS, etc, but that doesn't really help those of us trying to make do with the internet, for example. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax / t38 gateway
I'll look into those devices mentioned.
I think that I have one last question... I don't intend to have a
hardware fax machine on our end, I really just want it to get to
asterisk then email it from there. I know this can be done with
hylafax/iaxmodem etc, I actually have gotten that to work
intermittently, but supposing I did find an ITSP with t.38 support,
what would the steps be on the asterisk to receive that fax? I presume
it would just be something like:
exten = _XX.,n,Answer()
exten = _XX.,n,Wait(3)
exten = _XX.,n,Set(TIFFFILE=/var/spool/faxes/incoming-fax.tif)
exten = _XX.,n,ReceiveFAX(${TIFFFILE})
exten = _XX.,n,Set([EMAIL PROTECTED])
exten = _XX.,n,System('mewencode -e ${TIFFILE} | mail -s fax ${EMAIL}')
exten = _XX.,n,System('rm ${TIFFILE}')
That is what I was trying before I realized that it wouldn't work due
to ReceiveFAX() expecting t.38, right? But if it were coming in as t.
38 that is all there would be to it?
Thanks once again for taking the time to answer my questions.
Brendan Martens
On Oct 24, 2008, at 3:22 AM, Olivier wrote:
2008/10/23 Brendan Martens [EMAIL PROTECTED]
Indeed I am going for pure voip and trying to figure out how to
implement t.38, as you suggest.
On Oct 23, 2008, at 2:08 AM, Olivier wrote:
I think Brendan is asking about endpoints (how to connect fax
machines to pure VoIP).
Short answer:
- you could connect standalone T.38-enabled analog gateways to 1.4,
Like what? I'm not familiar with this tech, I googled around a bit but
didn't come up with much. I think I just don't know the lingo yet. :
( Could you point out one of these?
Linksys PAP2 or 3102 for instance
or Patton M-ATA
In fact, I would say most analog gateways with FXS port should also
support T.38.
In this case, your setup would be :
ISTP xDSL --- router ---LAN ---Asterisk 1.4 ---LAN ---
analog gateway === fax machine
As you mentioned, your IP Telephony Service Provider, would have to
provide T.30/T.38 conversion so that whenever you're sending or
receiving a fax, it would flow in ou or of your network.
- with 1.6, you can also use an analog board inside a server and
connect fax machines to this board.
So basically what you're saying is that to do this (convert the analog
to t.38) myself I would still need to have analog coming into my
asterisk server (which makes sense, but doesn't help me avoid paying
for normal phone lines)... Sounds to me like in this situation t.38
would be purely for getting faxes around on my own asterisk(s) if that
became necessary.
What I meant is that, instead of using a separate box for connecting
your own fax machine, you could use an analog board such as :
ITSP xDSL --- router ---LAN ---Asterisk 1.6 w/ FXS board
=== fax machine
Just as previous 1.4 setup, you wouldn't need a separate analog line
for faxing.
But judging from your question, I would add that it's not common to
find an ITSP able to deliver T.38 services (inbound or outbound).
And if you want to be able to switch from one provider to another,
or simply for simplicity, it's recommended practice to dedicate an
analog line to faxing.
You setup becomes :
ITSP xDSL --- router ---LAN ---Asterisk 1.6 w/ FXO-FXS
board === fax machine
|
|
PSTN ===
I should also add that if you're having a single fax machine, maybe
you should just connect it to an analog line.
Which leads me to my other question again, is there some sort of
internet service that will do the analog to t.38 conversion for me and
then pass the t.38 on to my asterisk server?
In your previous question you said pure VoIP which implied you had
found such provider.
Here you will some answers :
http://www.voip-info.org/wiki/view/VOIP+Service+Providers+T.38
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Andrew you mentioned something about sip providers that support t.38?
When you say support, do you mean that they have passthrough turned
on, or they will actually do an analog t.30 to t.38 conversion for
you? That may be what I'm after... If you, or anyone else, know of a
provider that does this could you point me in the right direction?
Thank you all for your thoughts.
Brendan Martens
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Re: [asterisk-users] fax / t38 gateway
Do you have any recommendations for good ones, or, non-buggy ones? Brendan Martens On Oct 24, 2008, at 7:48 AM, Steve Underwood wrote: Olivier wrote: Linksys PAP2 or 3102 for instance or Patton M-ATA In fact, I would say most analog gateways with FXS port should also support T.38. In this case, your setup would be : That list rather poorly supports your argument. The PAP2 and the PAP2T do *not* support T.38, despite numerous arguments you'll find to the contrary. Personally I believe Linksys, the manual, and the menus. Actually T.38 support is far from universal, and a lot of ATAs with support are as buggy as a roache nest Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax / t38 gateway
Indeed I am going for pure voip and trying to figure out how to implement t.38, as you suggest. On Oct 23, 2008, at 2:08 AM, Olivier wrote: I think Brendan is asking about endpoints (how to connect fax machines to pure VoIP). Short answer: - you could connect standalone T.38-enabled analog gateways to 1.4, Like what? I'm not familiar with this tech, I googled around a bit but didn't come up with much. I think I just don't know the lingo yet. : ( Could you point out one of these? - with 1.6, you can also use an analog board inside a server and connect fax machines to this board. So basically what you're saying is that to do this (convert the analog to t.38) myself I would still need to have analog coming into my asterisk server (which makes sense, but doesn't help me avoid paying for normal phone lines)... Sounds to me like in this situation t.38 would be purely for getting faxes around on my own asterisk(s) if that became necessary. Which leads me to my other question again, is there some sort of internet service that will do the analog to t.38 conversion for me and then pass the t.38 on to my asterisk server? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Andrew you mentioned something about sip providers that support t.38? When you say support, do you mean that they have passthrough turned on, or they will actually do an analog t.30 to t.38 conversion for you? That may be what I'm after... If you, or anyone else, know of a provider that does this could you point me in the right direction? Thank you all for your thoughts. Brendan Martens ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] fax / t38 gateway
I'm trying to figure out how to handle our fax line when we switch to our asterisk for voice. After a lot of reading and poking about I have concluded, as have many others it would seem, that the best thing to do is either to have a separate pstn fax line or use some sort of internet faxing service rather than try and make faxing work in a way it's not meant to over voip lines. The question I can't seem to find a good answer to is if there is a service/software that would allow a DID to be transferred to them and then they perform the t.38 gateway/conversion functions to which I can connect with asterisk as a t.38 endpoint and originator, or if there is a way that I could host that on my own server? So essentially I am a bit confused that asterisk supports t.38 as an endpoint or originator, but there doesn't seem to be a way to convert to/from analog for interoperating with normal fax machines. I'm sure something exists or the code wouldn't have been written into asterisk... Can someone point me in the right direction? Brendan Martens ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax / t38 gateway
I am using 1.6.0.1 and we are going to be pure voip. I know it has pass through and termination, but that is useless if I don't have a way to transform the analog t.30 to t.38 before it gets to me. That is where my confusion lays, is there some way of doing this that I am not aware of? Brendan Martens On Oct 22, 2008, at 3:02 PM, Jonn R Taylor wrote: What version of *? Are you going all VOIP for your voice or are you using a T1/E1? *? 1.4 has t38 pass-through and 1.6 has pass-through and termination, but 1.6 was just release and I would not suggest using it in a production environment unless you can tolerate problem or even outages. If you are planning on using a T1/E1 then send incoming calls to iaxmodem/hylafax or to an ATA/FXS card. Either works very well. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Brendan Martens Sent: Wednesday, October 22, 2008 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] fax / t38 gateway I'm trying to figure out how to handle our fax line when we switch to our asterisk for voice. After a lot of reading and poking about I have concluded, as have many others it would seem, that the best thing to do is either to have a separate pstn fax line or use some sort of internet faxing service rather than try and make faxing work in a way it's not meant to over voip lines. The question I can't seem to find a good answer to is if there is a service/software that would allow a DID to be transferred to them and then they perform the t.38 gateway/conversion functions to which I can connect with asterisk as a t.38 endpoint and originator, or if there is a way that I could host that on my own server? So essentially I am a bit confused that asterisk supports t.38 as an endpoint or originator, but there doesn't seem to be a way to convert to/from analog for interoperating with normal fax machines. I'm sure something exists or the code wouldn't have been written into asterisk... Can someone point me in the right direction? Brendan Martens ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemail issues with 1.6.0
I'm trying to get VoiceMailMain() to work properly, but it refuses. : ( I am using IMAP_STORAGE, which is functioning fine now... My voicemail.conf user line: 6000 = 1234,Brendan's Mailbox,,,[EMAIL PROTECTED]| imappassword=password 6000 = d,Brendan Martens My voicemail extension in extensions.conf: exten = 700,1,VoiceMailMain() And the output on the console: -- Executing [EMAIL PROTECTED]:1] VoiceMailMain(SIP/ 6000-489125a8, [EMAIL PROTECTED]) in new stack -- SIP/6000-489125a8 Playing 'vm-password.gsm' (language 'en') -- Incorrect password '1234' for user '6000' (context = default) -- SIP/6000-489125a8 Playing 'vm-incorrect.gsm' (language 'en') Very oddly I am having issues with updates to the files taking effect, I have changed the VoiceMailMain() to that after having previously set it to use a mailbox directly, but it still doesn't ask me what mailbox I want to check!? It seems to be stuck with mbox 6000... And before anyone asks, yes I have reloaded after making the changes, even a full restart does nothing. I am calling from ext 6000, but even when setting it to use [EMAIL PROTECTED] it still tries to check my 6000 mailbox. Regardless of some oddities in changes taking effect, the password/ user combination above is clearly correct I am baffled as to why it won't take the password, and why asterisk isn't finding my changes to the conf files. Brendan Martens ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
The reason for this is that 1.6.0 does not support dahdi. It was a mistake when it was listed as an included feature. The documentation for it has been removed in 1.6.0.1. If you need dahdi you need to go to 1.6.1. This is documented in this changelog: http://downloads.digium.com/pub/asterisk/ChangeLog-1.6.0.1 Brendan Martens On Oct 10, 2008, at 8:33 AM, Jim Duda wrote: Does anyone know what this error message means? Unable to create channel of type 'DAHDI' (cause 0 - Unknown) I've upgraded to 1.6.0 with dahdi 2.0. For some reason my outbound dahdi calls are not going through. At some point, it starts to work, but I don't know what the trigger is. Out of the blue, outbound calls start to work. I had been using asterisk-1.6-beta9 with zaptel without any problems. Thanks, Jim -- Executing [EMAIL PROTECTED]:1] Macro(SIP/111-b4e05610, dialout-dahdi,18005551212) in new stack -- Executing [EMAIL PROTECTED]:3] Set(SIP/111-b4e05610, DYNAMIC_FEATURES=outflash) in new stack -- Executing [EMAIL PROTECTED]:4] Dial(SIP/111-b4e05610, DAHDI/4/18005551212,40,tr) in new stack [Oct 10 08:29:10] WARNING[4365]: app_dial.c:1450 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:5] NoOp(SIP/111-b4e05610, Dial Status:CHANUNAVAIL) in new stack -- Executing [EMAIL PROTECTED]:6] Goto(SIP/111-b4e05610, s-CHANUNAVAIL,1) in new stack -- Goto (macro-dialout-dahdi,s-CHANUNAVAIL,1) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
I see, Thank you for the clarification. Brendan Martens On Oct 10, 2008, at 9:31 AM, Kevin P. Fleming wrote: Brendan Martens wrote: The reason for this is that 1.6.0 does not support dahdi. It was a mistake when it was listed as an included feature. The documentation for it has been removed in 1.6.0.1. If you need dahdi you need to go to 1.6.1. That is incorrect. There was one small feature (the 'dahdichan' configuration option, used when creating a new-style chan_dahdi.conf instead of the format used by zapata.conf) that was mistakenly included in the sample config file in 1.6.0, which has how been removed in 1.6.0.1. The dahdichan config option is supported in 1.6.1. 1.6.0 fully supports DAHDI, and does not support Zaptel at all. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.0.1 ??
You can find the changelog in the the downloads area: http://downloads.digium.com/pub/telephony/asterisk/ Excerpted from http://downloads.digium.com/pub/telephony/asterisk/ChangeLog-1.6.0.1 : 2008-10-08 Russell Bryant [EMAIL PROTECTED] * Asterisk 1.6.0.1 released. * configs/chan_dahdi.conf.sample: Remove mention of configuration sections for defining channels in chan_dahdi.conf. This code is in 1.6.1, and was not merged into 1.6.0. Brendan Martens On Oct 9, 2008, at 11:18 PM, sean darcy wrote: In download dated 10/9. Bug fix? Mistake? sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((
I assume you guys are using 1.6.0, yeah? Looks like there was some sort of confusion about dahdi in 1.6.0... I just saw this because of Sean Darcy's question about 1.6.0.1 in a different thread. This is from the 1.6.0.1 changelog: 2008-10-08 Russell Bryant [EMAIL PROTECTED] * Asterisk 1.6.0.1 released. * configs/chan_dahdi.conf.sample: Remove mention of configuration sections for defining channels in chan_dahdi.conf. This code is in 1.6.1, and was not merged into 1.6.0. Maybe that explains why dahdi won't work right? I don't know for sure, just noticed this and thought it may be applicable. Brendan Martens On Oct 9, 2008, at 11:00 PM, Tzafrir Cohen wrote: On Thu, Oct 09, 2008 at 08:12:46PM -0400, Alex Balashov wrote: Good point. I have a T100P that will not be seen by DAHDI for anything, but works fabulously with Zaptel. Which driver does it use? Is it shown by dahdi_hardware / zaptel_hardware ? (for zaptel_hardware: try in zaptel 1.4.12 and above) . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ldap usage in 1.6.0
Here, I've written a perl script that rewrites the actual sip.conf itself (as well as generates a custom myexten.conf file, which is included in the main extensions.conf file.) I was hoping to keep it all native to asterisk, but I would be willing to give that a try. Where can I get this script? Brendan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Creating Asterisk Binary Package
Jim Boykin wrote: I know about those packages. Questions is how do we use those packages to build our own RPM. We use asterisk SVN trunk. asterisk usually comes with asterisk.spec and make target rpm. With some slight modifications on the spec file you can pretty much build whatever you need into the package. You can try checkinstall. It makes a package (it supports a few kinds, rpm being one of them) out of the software you compiled. Basically instead of finishing with make install you just do checkinstall and it will make a package and then use your packaging system to install it. I use this often for Debian and it works very well there. You're distribution very likely has checkinstall available in it's main repository. If not the website is here for more info: http://www.asic-linux.com.mx/~izto/checkinstall/ Brendan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 or 1.6 ???
On Oct 6, 2008, at 3:52 PM, Gordon Henderson wrote: The answer you are looking for is that you should be using a supported, stable version, and right now, 1.4 is the only one that fits. If I were starting today, I'd go with 1.4. 1.6.0 has just been released. Personally I'd start with that because then you don't stuck with generation old features, and as you are just starting you aren't locked into any feature sets or syntax issues, etc. Of course as it has just been released there are undoubtedly some bugs yet to be discovered, 1.4 has been around a while and will probably be easier to find support/documentation for. Brendan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ldap usage in 1.6.0
Hello, I'm trying to figure out how to implement 1.6.0 with some ldap integration, but it's hard to figure out if I can do what I want. Basically I want to do only some lookup of values from ldap, as opposed to storing everything related to my sip users in ldap. For instance, would there be a way to lookup only certain context items from an ldap attribute in extensions.conf? Or in sip.conf? Something like this: user.conf [6000] hassip = yes hasiax = yes userfrom = ldapattribute insecure = route secret = anotherldapattribute type = friend context = ldapattrib3 It's looking to me like the way that ldap with 1.6.0 is meant to be used is more as a replacement for certain .conf files, like how odbc is used, and not really for referencing occasionally. I'm pretty new to asterisk so any guidance on this issue would be welcomed. Maybe if I explain a little overview of my end goal someone can help me more efficiently. I have an ldap directory on an OSX server, I want to create asterisk extensions for all of those users based on the extension, name, and password held in the ldap database. But I do not want to store whole .configs in ldap. Any ideas on how to go about this would be great. Brendan Martens ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ldap usage in 1.6.0
Thanks for the reply. Hmmm 1. I would provide Asterisk its own LDAP directory and synchronize it with entreprise directory as I think it should be simpler to synchronize 2 LDAP directories than coordinate Asterisk and Active Directory evolutions. This may work, but my end goal is really to simplify, not complicate. If I can't get the information I need for sip users etc from ldap then I'll just have to skip it... I need to not be the only person that can manage whatever setup I end up with. : ( 2. IMHO, many people are confusing SIP secrets (from sip.conf) which somehow authenticate hardware with user passwords which authenticate persons. I wouldn't try to make those 2 values equal. Hmm, once again with the integration and the simplifying, one of the biggest reasons I want access to ldap is to be able to authenticate there, I really don't want to introduce another place to manage authentication. Most of my users will be using sip phones and I don't want to give them another user/password combo to remember. : ( 3. Asterisk's LDAP directory should be the reference for anything related to telephony. Changes could be automatically propagated from Asterisk to corporate directory. 4. Corporate directory should be the reference for user management. Changes should be manually propagated from corporate directory to Asterisk as I don't believe it could be easy to allocate nor free telephony resources whenever a user is created or deleted in corporate directory. Not quite sure I follow here... If a user was deleted from my ldap directory, the corresponding sip phone should fail registration, right? Having thought some more about my issue I think I can perhaps ask my question more succinctly: is it possible to get dynamic (or realtime) data from ldap within the various .conf files? If there is not a convenient function for getting this in the .conf files, what if I somehow specified a global variable within the res_ldap.conf and referenced that value inside the other .conf files? Is this possible? Sorry if these are very basic questions, I just haven't been able to find answers to them. : ( Brendan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
