Koozali.org: home of the SME Server
Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: SARK devs on October 24, 2006, 04:14:46 PM
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HI all
-344 went up onto our ftp server this afternoon.
ftp://81.149.154.14/Pre-Releases/
Routes, queues and agents have all been updated to better show what's actually going on in real time - you can see what agents are logged on to which queues and stuff like that. The Routes panel has been updated to reflect the dial plans making it easier to compare plans.
There is also a cute paging feature which works over infinite page groups that we've just done for a biggish customer here in the UK. You define a page group in aliases and fire up the page operation with *40*nnnn (where nnnn is the alias). *40* (no alias) pages everything.
The operator extension is assumed to be privileged so the page will go straight thru. From all other extensions you will be required to enter a password.
We've tested paging on Snoms, Aastra's, Linksys/Sipura's, and Grandstreams. They all work fine but the grandstreams (101/102) need the "auto-answer" switch set to on.
Kind Regards
Selintra
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Hi Jeff/Selintra,
I've got a HFC-SDN card installed with SAIL 344 and i'm making a call with X-lite to my own mobile, but when i end the call from X-lite it is not beeing hung up:
-- Executing AGI("SIP/5000-c9b1", "selintra|OutCluster|0666666666") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script selintra completed, returning 0
-- Executing AGI("SIP/5000-c9b1", "selintra|OutRoute|ISDN_Out") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script Executing Application: (Dial) Options: (Zap/g2/0666666666)
-- Requested transfer capability: 0x00 - SPEECH
-- Called g2/0666666666
-- Zap/1-1 answered SIP/5000-c9b1
Oct 24 18:16:22 WARNING[3249]: chan_zap.c:8503 zt_pri_error: 1 updating callstate, peercallstate 2 to 1
-- PROGRESS with cause code 100 received
== Forcing Marker bit, because SSRC has changed
Oct 24 18:16:37 NOTICE[3443]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.1.71
-- Hungup 'Zap/1-1'
== Spawn extension (default, 0666666666, 1) exited non-zero on 'SIP/5000-c9b1'
-- Executing Hangup("SIP/5000-c9b1", "") in new stack
== Spawn extension (default, h, 1) exited non-zero on 'SIP/5000-c9b1'
cyclops*CLI>
Regards,
jester.
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Hi Jester
I'm really not sure what you mean.
but when i end the call from X-lite it is not beeing hung up:
What is not being hungup? The near end or the far end?
The hangup code looks fine in the log.
By the way, you may want to turn of silence suppression on the x-lite - asterisk doesn't like it.
Kind regards
Selintra
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Hi Selintra,
If the X-lite is hung up first, my mobile keeps ringing... if i then ignore this call it is beeing cought by my voicemail and even records an message (silence).
If i ignore the call on my mobile first i can hear my mobile's voicemail starting on the X-lite client. Even if i then hang up the X-lite before my mobile's voicemail start blabbering about leaving a message after the beep it still records a message.
So It looks like the call is actually NOT beeing hung up. Anything i can test or provide you with?!
Regards,
jester.
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Hi Jester
This is almost certainly NOT a SAIL problem. We don't do anything that low down in the code. Instead it looks like a call progress issue with BriStuff or Zaptel.
What happens if you do the same experiment to an ordinary phone (instead of your mobile)? Same problem?
Kind Regards
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Hi Selintra,
It's the same with a normal phone
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Hi Jester
Take a look at this thread...
http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=4958&forum=2
Looks like a bug in libpri.
We'll schedule a recompile for you but this will be the last for Bristuff. Digium haven't gone with it for their BRI card and it is just not stable enough to put into a customer site.
Kind Regards
Selintra
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Selintra,
Are there any other BRI cards (AVM, Beronet, Eicon, Fritz, Sangoma) going to be supported by SAIL/Selintra ?! Or maybe a different option like the ChanCapi module?
Regards,
jester.
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Are there any other BRI cards (AVM, Beronet, Eicon, Fritz, Sangoma) going to be supported by SAIL/Selintra ?! Or maybe a different option like the ChanCapi module?
Hi Jester
BRI support in Asterisk is a mess. Three or four different driver implementations several different hardware offerings and none of them, in our opinion, good enough to justify the effort needed to get them up and running and properly supported in a production environment.
So, for our commercial customers, I've avoided the whole issue and used purpose-built VoIP/ISDN gateways. They aren't cheap but they work properly in real-world environments and they don't break. Initially we have deployed, and had very good results with, MultiTech units but we are looking to evaluate others.
Kind Regards
Selintra
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I have a few questions that I know are a bit off-topic for this thread, but I didn't want to start yet another thread. I downloaded the 344 build, now I would like to make sure I have the correct RPMS. I downloaded the ones here - http://mirror.contribs.org/smeserver/contribs/selintra/RPMS/AsteriskForSail-2.1.13/None-ISDN/, there are older versions here - http://mirror.contribs.org/smeserver/contribs/selintra/RPMS/, but there is also a sounds rpm there as well. Do, I need that? I have procured me a TDM400P, which I can use, and a TE405P.
I have a bit of a learning curve here, is the SARK/SAIL 2.1.14 documentation valid for what I have downloaded? If I use this documentation, should I still get the Asterisk docs as well? Do either of the guides have a phone compability list? Any other noob answers you want to give me to quesitons that I didn't ask, but should have? :lol:
Thanks for the time,
John Bennett
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Hello John,
You have downloaded the correct rpms but you do also need the sounds pack.
The documentation is valid for what you have downloaded. There are a few new features in 344 which we haven't updated on the docs site yet but you are pretty much good-to-go.
Third party Asterisk documentation is an advantage. We can recommend the O'Reilly book on the subject - it is ISBN 0-596-00962-3. SAIL doesn't attempt to hide Asterisk from the user it just acts like a framework, enforcing good practice, so it is still advantageous to know what's going on under the covers.
Phone compatibility - SARK already knows about most popular phones and can automatically define and provision them or you. For more unusual, or newer, phones we have a general purpose phone type that you can modify to suit.
To set-up your TDM400 follow the guide in appendix 1.2 of the documentation.
SARK doesn't currently support Digium PRI cards like your 405. However, we'd be more than happy to work with you to get it running. The theory is pretty straightforward and there are lots of PRI/Asterisk installations out there, we've just never had occasion to do it.
Other than that, nothing springs to mind except to say that it's fun fooling around with computer-driven phones.
:-)
Kind Regards
jeff@selintra
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John, you will find some useful information about asterisk at:
http://www.voip-info.org/tiki-index.php?page=Asterisk
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You have downloaded the correct rpms but you do also need the sounds pack.
Jeff, why is this sound pack necessary, the asterisk rpm found on your site already have sound in-built, albeit US voices?
Regards, chris.
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Hi Chris,
I meant the smeserver-asterisk-sounds rpm. It is an amalgam of the base sound library plus the asterisk-addon-sounds library.
So, to clarify, - you should install...
smeserver-asterisk-1.2.10-1.i686.rpm
smeserver-asterisk-zappri-MPP-1.2.6-1.i686.rpm
smeserver-asterisk-sounds-1.2.2-2.noarch.rpm
This will give you 1.2.10 asterisk with the full US sounds pack.
Kind Regards
Jeff
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...
Third party Asterisk documentation is an advantage. We can recommend the O'Reilly book on the subject - it is ISBN 0-596-00962-3. SAIL doesn't attempt to hide Asterisk from the user it just acts like a framework, enforcing good practice, so it is still advantageous to know what's going on under the covers.
To set-up your TDM400 follow the guide in appendix 1.2 of the documentation.
SARK doesn't currently support Digium PRI cards like your 405. However, we'd be more than happy to work with you to get it running. The theory is pretty straightforward and there are lots of PRI/Asterisk installations out there, we've just never had occasion to do it.
Thanks for replying. I did find that O'Reilly book as a free download. I can't really use the PRI card at home, I just found the cards at work and brought them home to play with. Since I don't have a FXO daughter card, 4 fxs's, I can't hook it up to my home line, so it looks like all I'll be able to do is set it up and test internally. I guess that may not really be true. I could probably call one of my IP phones at work, but it wouldn't be able to call me. I guess I'll hopefully soon see.
I may start a new thread on how people are using this at home, and get very familiar with the Asterisk sites.
Thanks again,
John
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So, to clarify, - you should install...
smeserver-asterisk-1.2.10-1.i686.rpm
smeserver-asterisk-zappri-MPP-1.2.6-1.i686.rpm
smeserver-asterisk-sounds-1.2.2-2.noarch.rpm
This will give you 1.2.10 asterisk with the full US sounds pack.
Kind Regards
Jeff
Does the documentation or Quickstart guides on your web site mention the sounds rpm ? Is this a required step for a fresh install ?
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Does the documentation or Quickstart guides on your web site mention the sounds rpm ?
Reading it back, no it doesn't and it should - we'll fix it.
Is this a required step for a fresh install ?
It isn't required in the sense that asterisk will run quite happily without it and,for example, for SARK/SAIL in the UK we use british voice prompts.
However, if you don't install it then you will get silences where prompts should be.
It is a noarch rpm and it can be installed at any time so you can put it on retrospectively.
Kind Regards
Selintra
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There is also a cute paging feature which works over infinite page groups that we've just done for a biggish customer here in the UK. You define a page group in aliases and fire up the page operation with *40*nnnn (where nnnn is the alias). *40* (no alias) pages everything.
This one got me... Just upgraded to 344 and tried *40* from receptionist, to be told that this is not a valid conf number. The problem is I do not have ztdummy or a zaptel card loaded. Page uses the meetme functionality which means that it needs a timing source. As I do not have a zaptel card loaded, remedy is:
Stop asterisk and do: modprobe ztdummy
(edited) - restart Asterisk
To make it stick on next reboot, go into the PCI cards panel and do a modprobe
Now its working as intended, great feature!
Thanks to Jeff for guidance.
chris
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Are there any other BRI cards (AVM, Beronet, Eicon, Fritz, Sangoma) going to be supported by SAIL/Selintra ?! Or maybe a different option like the ChanCapi module?
Hi Jester
BRI support in Asterisk is a mess. Three or four different driver implementations several different hardware offerings and none of them, in our opinion, good enough to justify the effort needed to get them up and running and properly supported in a production environment.
So, for our commercial customers, I've avoided the whole issue and used purpose-built VoIP/ISDN gateways. They aren't cheap but they work properly in real-world environments and they don't break. Initially we have deployed, and had very good results with, MultiTech units but we are looking to evaluate others.
Kind Regards
Selintra
Hi Selintra,
I fully agree that ISDN ( let say 'euro-isdn' ) support remains an issue within the * world, but it is also a major concern ( at least for the european market ) to improve that situation.
I don't know exactly on which market you are focussing, but , to my opinion ( not more than that ) this is a must for SMB.
To be honnest, with the exception of some quick-solved bugs, I never had 'stability' problems using Junghanns cards/software.
Many thanks for what you are doing !.
Herve