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Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: ntblade on February 22, 2007, 02:18:24 PM

Title: SAIL - Which version please?
Post by: ntblade on February 22, 2007, 02:18:24 PM: Hi,
I'm about to install SAIL again on a clean 7.1 server.
Please, what RPM versions should I install? (No ISDN)
What SME updates are safe with SAIL?
What kernel version should I be running? Is SMP OK?

Thanks
Norrie
Title: Re: SAIL - Which version please?
Post by: hervep on February 22, 2007, 03:19:00 PM: Quote from: "ntblade"
Hi,
I'm about to install SAIL again on a clean 7.1 server.
Please, what RPM versions should I install? (No ISDN)
What SME updates are safe with SAIL?
What kernel version should I be running? Is SMP OK?

Thanks
Norrie

I would suggest you to follow http://81.149.154.14/docs/cgi-bin/view

Be patient, looks to be down for now.

Best,

herve
Title: SAIL - Which version please?
Post by: SARK devs on February 22, 2007, 08:08:48 PM: OK...

Please read the notes as Herve suggests. Then...

Asterisk rpms here

http://mirror.contribs.org/smeserver//contribs/selintra/RPMS/AsteriskForSail-2.1.13/None-ISDN/

Use the latest versions of both zappri-MPP and smeserver-asterisk.

Sound rpms here (non-US)

http://mirror.contribs.org/smeserver//contribs/selintra/RPMS/languagepacks/

and here (US)

http://mirror.contribs.org/smeserver//contribs/selintra/RPMS/

SAIL itself is here...

ftp://81.149.154.14/Pre-Releases/

Latest release is .425

Quote
What SME updates are safe with SAIL?
What kernel version should I be running? Is SMP OK?

We now run full 7.1, however you do have to make a small change for each different kernel release. You can read about it here...

http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter2510

Any problems just give us a shout.

There are a lot of releases and it can be confusing. If in doubt, put the very latest release of each rpm up and you can't go far wrong!

Best

Selintra
Title: SAIL - Which version please?
Post by: ntblade on February 22, 2007, 09:59:18 PM: Thanks for the replies.

[root@test ~]# uname -r
2.6.9-42.0.3.EL
[root@test ~]# rpm -q selintra-sail
selintra-sail-2.1.14-425
[root@test ~]# rpm -q smeserver-asterisk
smeserver-asterisk-1.2.10-3
[root@test ~]# rpm -q smeserver-asterisk-zappri-MPP
smeserver-asterisk-zappri-MPP-1.2.6-1
[root@test ~]#

I'm confused. I've not done any updates so I'm running a stock 7.1 kernel. The documentation here:
http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter2510
goes up to 2.6.9-42.0.2.EL (is this a typo?) and says to run the 2.6.9.34.EL kernel but this isn't on my system so option 2 is what I'll try.

N
Title: SAIL - Which version please?
Post by: SARK devs on February 22, 2007, 10:11:11 PM: Hi

You are quite correct. 42.0.2 was the "latest" kernel when the notes were last revisited. 42.0.3 is the 7.1 kernel, but the same principle applies. Just copy the extra sub-directory into the 42.0.3 kernel directory and off you go.

Thanks for pointing this out - I have updated the docs page.

Kind Regards

Selintra
Title: SAIL - Which version please?
Post by: ntblade on February 23, 2007, 12:00:46 AM: Thanks for letting me know. Still no joy though. Should the zap dummy module still be loaded if no PCI cards are present?
Also, my SIP phone aren't showing as being connected even though *56* and *52* work OK.

Off to bed, will update in the morning.

N
Title: SAIL - Which version please?
Post by: SARK devs on February 23, 2007, 01:07:06 AM: Quote
Should the zap dummy module still be loaded if no PCI cards are present?

Yes it should. Asterisk gets its timing from zaptel. In the absence of any physical cards then SAIL will load ztdummy which acts as a timing source. Without it, things like conference rooms won't work.

Re phones....

Check your phones have registered correctly with "sip show peers" at the asterisk console. Usually if *56* works then they should be OK. If not, then it's usually a registration issue.

Kind Regards

Selintra
Title: SAIL - Which version please?
Post by: ntblade on February 23, 2007, 09:46:24 AM: I'm using the same hardware as in this:
http://forums.contribs.org/index.php?topic=34731.0
which after some help was working fine.

Quote from: "selintra"
Yes it should. Asterisk gets its timing from zaptel. In the absence of any physical cards then SAIL will load ztdummy which acts as a timing source. Without it, things like conference rooms won't work.

Asterisk doesn't start automatically, Dummy module never gets loaded.
Added an X100p clone and followed my previous post and the card gets detected.

Quote from: "selintra"
Re phones....
Check your phones have registered correctly with "sip show peers" at the asterisk console. Usually if *56* works then they should be OK. If not, then it's usually a registration issue.

Code: [Select]
test*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 5002/5002 (Unspecified) D 0 UNKNOWN 5001/5001 192.168.112.248 D 5060 UNREACHABLE 5000/5000 192.168.112.250 D 5060 UNREACHABLE 3 sip peers [0 online , 3 offline]
5002 is a softphone.

Back in a while
N
Title: SAIL - Which version please?
Post by: ntblade on February 23, 2007, 10:57:21 AM: From the CLI...

-- Registered SIP '5001' at 192.168.112.248 port 5060 expires 180
-- Saved useragent "TMS320VC50005.0.X.02.0.9.11" for peer 5001
Feb 23 09:42:43 NOTICE[5034]: chan_sip.c:11564 sip_poke_noanswer: Peer '5001' is now UNREACHABLE! Last qualify: 1
-- Registered SIP '5001' at 192.168.112.249 port 5060 expires 160
-- Saved useragent "SJphone/1.60.289a (SJ Labs)" for peer 5001
Feb 23 09:44:30 NOTICE[5034]: chan_sip.c:9856 handle_response_peerpoke: Peer '5001' is now REACHABLE! (1ms / 3000ms)
-- Registered SIP '5001' at 192.168.112.248 port 5060 expires 180
-- Saved useragent "TMS320VC50005.0.X.02.0.9.11" for peer 5001
Feb 23 09:45:23 NOTICE[5034]: chan_sip.c:11564 sip_poke_noanswer: Peer '5001' is now UNREACHABLE! Last qualify: 1
-- Registered SIP '5001' at 192.168.112.249 port 5060 expires 160
-- Saved useragent "SJphone/1.60.289a (SJ Labs)" for peer 5001
Feb 23 09:47:10 NOTICE[5034]: chan_sip.c:9856 handle_response_peerpoke: Peer '5001' is now REACHABLE! (1ms / 3000ms)
-- Registered SIP '5001' at 192.168.112.248 port 5060 expires 180
-- Saved useragent "TMS320VC50005.0.X.02.0.9.11" for peer 5001
Feb 23 09:48:03 NOTICE[5034]: chan_sip.c:11564 sip_poke_noanswer: Peer '5001' is now UNREACHABLE! Last qualify: 1
-- Registered SIP '5001' at 192.168.112.249 port 5060 expires 160
-- Saved useragent "SJphone/1.60.289a (SJ Labs)" for peer 5001
Feb 23 09:49:50 NOTICE[5034]: chan_sip.c:9856 handle_response_peerpoke: Peer '5001' is now REACHABLE! (1ms / 3000ms)
-- Registered SIP '5001' at 192.168.112.248 port 5060 expires 180
-- Saved useragent "TMS320VC50005.0.X.02.0.9.11" for peer 5001
Feb 23 09:50:43 NOTICE[5034]: chan_sip.c:11564 sip_poke_noanswer: Peer '5001' is now UNREACHABLE! Last qualify: 1
test*CLI>

Does this look like a network problem?
Title: SAIL - Which version please?
Post by: SARK devs on February 23, 2007, 11:14:57 AM: Hi NT

You are almost certainly spoofing the phone with the wrong return address in the SIP packets. This is why it registers and then immediately loses contact.

Look at the field "Your External IP Address: " in globals.

We usually see this error when that setting is incorrect.

Kind Regards

Selintra
Title: SAIL - Which version please?
Post by: hervep on February 23, 2007, 11:17:58 AM: Hi Ntblade,

Looks like 'sjphone' was trying to login simultaneously from two different IP addresses ...

Any IP conflict or LAN + WLAN activated on same pc ... ???

Quote
'5001' at 192.168.112.248

'5001' at 192.168.112.249

Best,

Hervé
Title: SAIL - Which version please?
Post by: ntblade on February 23, 2007, 11:53:15 AM: Yes you're right but even without sjphone connected I'm having the same problems.
I'm performing a fresh install of SME and I'll log everything I do and post the results. I suppose the first thing would be to get the dummy module to load?

N