Topic: [announce SAIL-2.1.15 and Asterisk-1.4 rpms]

[SARK devs]

Hello everyone,

We have a few new rpms for you to try.  

There is a new version of SAIL, which we have designated 2.1.15, together with the very latest version of Asterisk(1.4.1) and zaptel (1.4.0).  It also has a full mISDN channel implementation included, which should allow us to dispense with separate rpms for ISDN and none-ISDN users.

Don't try to install 2.1.15 with asterisk 1.2, it won't work (the rpm pre-reqs will stop you anyway).

All seems stable but we wouldn't recommend you put this stuff into production without fully testing the features you wish to use.

As you may know, 1.4 is quite different from Asterisk 1.2.  There are many new features and a lot of old stuff has disappeared. Fortunately, the AGI insulates SAIL from a lot of that.  However, before we crow too much about our pretty architecture, we'll wait for you all to give the new release a shake down.  I'm sure you'll find errors but that's OK, - we'll fix 'em.  

You can find the Asterisk rpms here

http://mirror.contribs.org/smeserver//contribs/selintra/RPMS/Asterisk-1.4.1/

Sail rpm is here

ftp://81.149.154.14/Pre-Releases/selintra-sail-2.1.15-443.noarch.rpm

You can simply yum localinstall or rpm -Uvh the new rpms over whatever you have now.  Only caveat is that you MUST run console-save after install and then you MUST open globals and do a commit in order to set up the etc/asterisk data sets.

Lastly, we have not finished regression testing this release so you should consider it as an alpha.  In other words, run it on a spare box for now because it just might break.

Over the next few weeks we will begin to exploit some of the cool new features in 1.4 so keep watching as we roll these out.

Kind Regards

Selintra

[Tib]

Install went well as far as I can tell on my test server.

Even did a server reboot and everything came online as it should.

Did a quick check on all settings and all seems to be fine.

My setup only has Astratel setup as the main trunk .... I did a few test calls to my home astratel number and there are no calls registering in CRD Database.

Also tried calling from home to work using voip numbers and still no logs in CDR database.

So far that all I could find.

Regards,

Tib

[SARK devs]

Thanks for the input Tib.

You do have records in /var/log/cdr-csv tho' - right?

Fix shortly for mysql CDR

Best

J

[Tib]

hmm ok ... I do not have anything in /var/log/cdr-csv

The file does not even exist.

Maybe I will try this out on my home machine and see what happens.

Looking at phpmyadmin I do have a cdr table in asterisk ... but 0 records.

Maybe it's just a past problem on this machine.

I don't remember checking to see if cdr had anything in it with the previous vertions.

Regards,

Tib

[Tib]

ok ...  I loged into my home machine from work ... and it doesn't have that file either but it shows up all the calls on the cdr page.

OK as as was typing this I noticed the asterisk area ... /var/log/asterisk/cdr-csv ..... now there are entries in that file.

Both my home server and the test server here have entries in the file.


Regards,

Tib

[SARK devs]

Oops!

Sorry Tib

I meant /var/log/asterisk/cdr/csv

   


Thanks again mate.

Best

J

[sonoracomm]

Sorry if this is a stupid question...

Is there a new Asterisk Sounds RPM also?  I think the sounds are somewhat different for 1.4 also, or no?

Thanks again,

G

[SARK devs]

HI

All of the base sounds are incuded in the 1.4 asterisk rpm.  These are USA English only at the moment.  If you want other languages/dialects you will need to download and add them manually.

There are differences in the sound directories (according to the 1.4 release notes) but you can add the UK english sound pack rpm and it just seems to work (at least, it did for us).

Hope this helps

Selintra

[sonoracomm]

Hi Jeff,

I tested the upgrade on my home (test) system last night.  Here are the steps I needed:
rpm -e smeserver-asterisk-sounds-1.2.2-2

Code: [Select]

rpm -Uvh http://mirror.contribs.org/smeserver//contribs/\
selintra/RPMS/Asterisk-1.4.1/smeserver-asterisk-zappri-MPP-1.4.0-4.i686.rpm

rpm -Uvh http://mirror.contribs.org/smeserver//contribs/\
selintra/RPMS/Asterisk-1.4.1/smeserver-asterisk-1.4.1-3.i686.rpm

rpm -Uvh ftp://81.149.154.14/Pre-Releases/selintra-sail-2.1.15-443.noarch.rpm

signal-event console-save

ln -s /etc/rc.d/init.d/e-smith-service /etc/rc.d/rc7.d/S93asterisk

service asterisk restart

signal-event reboot
Everything seemed to go well, but I have one (big) problem.

I am using a SPA-2100 ATA (ext. 5000) and a Telasip trunk.

I have a situation that is similar to one-way audio...but not quite.  Audio seems fine inbound, but outbound there is either no audio at all or a terribly delayed, crackly, faint, completely unusable audio heard on the cell phone.

The console output looks OK to me...

Code: [Select]

[root@sol ~]# asterisk -vvvvvvvvvvvr
Asterisk 1.4.1, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.1 currently running on sol (pid = 4442)
Verbosity is at least 11
    -- Remote UNIX connection
    -- Executing [6611293@internal:1] AGI("SIP/5000-08abc0f8", "selintra|OutCluster|6611293") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script selintra completed, returning 0
    -- Executing [6611293@default:1] AGI("SIP/5000-08abc0f8", "selintra|OutRoute|Primary SIP") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Set) Options: (CALLERID(NUMBER)=5203997467))
    -- AGI Script Executing Application: (Dial) Options: (SIP/15206611293@Telasip)
    -- Called 15206611293@Telasip
    -- Call on SIP/Telasip-08ac2ef8 left from hold
    -- SIP/Telasip-08ac2ef8 is making progress passing it to SIP/5000-08abc0f8
    -- Call on SIP/Telasip-08ac2ef8 left from hold
    -- SIP/Telasip-08ac2ef8 answered SIP/5000-08abc0f8
    -- Packet2Packet bridging SIP/5000-08abc0f8 and SIP/Telasip-08ac2ef8
  == Spawn extension (default, 6611293, 1) exited non-zero on 'SIP/5000-08abc0f8'
    -- Executing [h@default:1] Hangup("SIP/5000-08abc0f8", "") in new stack
  == Spawn extension (default, h, 1) exited non-zero on 'SIP/5000-08abc0f8'
sol*CLI>

Any tips for me?

Thanks in advance,

G

p.s.  I am NOT auto-provisioning the SPA-2100 so here is the config:

Code: [Select]


[5000]
type=friend
username=XXXX
secret=XXXX
mailbox=5000
host=dynamic
qualify=3000
context=internal
callerid="grchome" <5000>
canreinvite=no
pickupgroup=1
callgroup=1
disallow=all
allow=alaw
allow=ulaw

[sonoracomm]

Also, there are 31 Asterisk processes spawned.  Is that normal?

I enabled the syslog functionality on the server and pointed the SPA-2100 at it.  I also enabled debug mode on the SPA-2100.  Here was the output upon reboot and making a call:

Code: [Select]

Mar 23 09:18:31 ata fu:0:39d8, 03cc 0001
Mar 23 09:19:04 ata [0]Off Hook
Mar 23 09:19:07 ata 2. Report digit 6 (1)(40 ms)
Mar 23 09:19:07 ata 2. Report digit 6 (1)(40 ms)
Mar 23 09:19:07 ata 2. Report digit 1 (1)(40 ms)
Mar 23 09:19:07 ata 2. Report digit 1 (1)(40 ms)
Mar 23 09:19:07 ata 2. Report digit 2 (1)(40 ms)
Mar 23 09:19:08 ata 2. Report digit 9 (1)(40 ms)
Mar 23 09:19:08 ata 2. Report digit 3 (1)(40 ms)
Mar 23 09:19:20 ata Calling:6611293@192.168.2.1:0
Mar 23 09:19:20 ata [0:0]AUD ALLOC CALL (port=16388)
Mar 23 09:19:20 ata [0:0]RTP Rx Up
Mar 23 09:19:32 ata [0:0]RTP Rx 1st PKT @16388(2)
Mar 23 09:19:32 ata [0:0]ENC INIT 0
Mar 23 09:19:32 ata [0:0]RTP Tx Up (pt=0->c0a80201:16100)
Mar 23 09:19:32 ata [0:0]RTCP Tx Up
Mar 23 09:19:32 ata CC:CallProgress
Mar 23 09:19:32 ata [0:0]DEC INIT 8
Mar 23 09:19:32 ata [0:0]DEC INIT 0
Mar 23 09:19:37 ata [0:0]RTP Tx Dn
Mar 23 09:19:37 ata [0:0]ENC INIT 0
Mar 23 09:19:37 ata [0:0]RTP Tx Up (pt=0->c0a80201:16100)
Mar 23 09:19:37 ata CC:Remote Resume
Mar 23 09:19:37 ata CC:Connected
Mar 23 09:19:37 ata RTP:SSRC changed 758c3a43->100e27ee
Mar 23 09:19:37 ata RTP:SSRC changed 100e27ee->9ac06066
Mar 23 09:19:37 ata [0:0]RxBigGapSeqNo:9557 55104
Mar 23 09:19:53 ata [0]On Hook
Mar 23 09:19:53 ata [0]FM Alert Stop RxTx (c=0022c260;a=0)
Mar 23 09:19:53 ata [0:0]AUD Rel Call
Mar 23 09:19:53 ata DLG Terminated 2a750c
Mar 23 09:20:09 ata Sess Terminated
Mar 23 09:20:26 ata CC:Clean Up
Mar 23 09:20:26 ata --- OBJ POOL STAT ---
Mar 23 09:20:26 ata OP:RTPRXB =  96 ( 96  192)  
Mar 23 09:20:26 ata OP:RTPREB =  40 ( 40   48)
Mar 23 09:20:26 ata OP:RTPTXB =  64 ( 64  108)  
Mar 23 09:20:26 ata OP:TIMEOU = 108 (120   52)
Mar 23 09:20:26 ata OP:SIPCOR =   0 (  1   28)  
Mar 23 09:20:26 ata OP:SIPCTS =  32 ( 32  568)
Mar 23 09:20:26 ata OP:SIPSTS =  30 ( 32 3492)  
Mar 23 09:20:26 ata OP:SIPAUS =   0 (  8  588)
Mar 23 09:20:26 ata OP:SIPDLG =  10 ( 10  148)  
Mar 23 09:20:26 ata OP:SIPSES =  12 ( 12 8200)
Mar 23 09:20:26 ata OP:SIPREG =   2 (  4  292)  
Mar 23 09:20:26 ata OP:SIPLIN =   0 (  2  140)
Mar 23 09:20:26 ata OP:SUBDLG =   2 (  2 6436)  
Mar 23 09:20:26 ata OP:STUNTS =  16 ( 16   68)
Mar 23 09:20:26 ata  
Mar 23 09:21:16 ata [0]RegOK. NextReg in 177 (1)
Mar 23 09:21:16 ata [1]RegOK. NextReg in 177 (1)
Mar 23 09:21:23 ata [1]MWI 1 2/0
It didn't help me, but...

Thanks again,

G

[SARK devs]

Hi G

Not a clue why you have a problem with the ATA.  We have the same code (1.4 and 443) in production on the office switch and it isn't giving any sound problems.  However, we have no ATA's on it.  We have 3 Aastra 9112i's, a Snom 360, Snom 300, a Polycomm (don't remember the model), a Mitel 5215,  various Grandstreams (101s and GXS2000's) and a Siemens C460 SIP/DECT unit and none of them are currently giving any sound problems going out over a mixture of UK VoIP carriers (Gamma, Comms.com, Gradwell & Voiceflex).

So... Don't quite know how to advise you.  Do you have another SIP device you can try?  Maybe that might help isolate the problem area.

Kind Regards

Selintra

[jazbokes]

Hi,

Will u be posting the asterisk-1.4.1 and zappri-1.4.0 srpms  

Thanks in advance..

[SARK devs]

Will u be posting the asterisk-1.4.1 and zappri-1.4.0 srpms

Yes of course, just haven't gotten 'round to it yet.  It takes a long time to upload them from our test server because the line it's on doesn't have much uplift capability.  Also, the rpms still need a little teaking so we will upload SRPMS when we're happy that they are stable.  That way we only have to do it once.



Kind Regards

Selintra

[JonB]

A couple of things.

To get music on hold working you need to modify musiconhold.conf and change

directory=>/var/lib/asterisk/mohmp3

to

directory=>/var/lib/asterisk/moh

To get other sounds working you need to add to sip.conf and iax.conf

language=xx where xx equals the country e.g for NZ

Install sme-ast-en-nz-gpl-sounds-1.0.0-1.noarch.rpm

then add

language=nz

to sip.conf and iax.conf

Jon

[SARK devs]

Hi Jon,

Thanks for this,

We missed the moh - we just installed over a previous asterisk version so mohmp3 was still there and still works.

I'll mod the database to suit on the next out.

Re sounds - it currently defaults to gb in sip and iax.conf - which is a bit naughty.

btw if you are using zap lines you should also add language= to zaptel.

Also, there are some other new features you might like to check out, like jitter buffers in sip and so forth.

It's kind of a tuning process at the moment.

Thanks for your input mate.

Best

J