Topic: SAIL Vs Asterisk@Home

[Franco]

What are your experiences with these 02 contribs?
I just started migrating to SME7 ( New Machine   ) and I'm planning on trying the WildFire contrib as well.
I used asterisk@home before but I noticed some real good comments on SAIL.

Thanks,

[WillKemp]

From what i've heard, A@H is considerably better than it used to be - now it uses FreePBX, rather than AMP. It's also about to have a name change.

I'd like to give A@H a try sometime, to be able to assess the significant differences between it and SAIL, but i doubt that's likely to happen in the near future. If you've used A@H since it switched from AMP to FreePBX, maybe you could install SAIL and then you can tell us how they compare.

[Franco]

@WillKemp,
Thanks for repplying!
It is needless to say that when I mentioned 'but I noticed some real good comments on SAIL' I was referring to you
I have not tried the new version of @home with FreePBX.
I'll install SAIL tonight, whatta heck! Then install the new @home after and compare both (clean system of course).
I'll let you know how it goes.

Thanks,

[Franco]

Ok, so I went ahead with the SAIL installation, so far I'm impressed!
Couple of things thou:
1- Voicemail does not work?
2- I didn't have to open ports on previous installs, I'm testing against FWD and I can see that I'm registered with them:

Code: [Select]

4613XX/4613XX              69.90.155.70                5060     OK (347 ms)
2123/2123                  172.16.0.150     D          5060     OK (172 ms)
but I can't make/receive calls
3- No ring group?

Thanks,

[WillKemp]

1- Voicemail does not work?

I'm not sure about this - i use a "custom app" for my incoming calls, because i had a bit of a "non-standard" setup in my old installation, which i transferred to this one, so i haven't checked it out. But i'm pretty sure voicemail should work to some extent.

2- I didn't have to open ports on previous installs, I'm testing against FWD and I can see that I'm registered with them:

Code: [Select]

4613XX/4613XX              69.90.155.70                5060     OK (347 ms)
2123/2123                  172.16.0.150     D          5060     OK (172 ms)
but I can't make/receive calls

What happens when you try? Get an Asterisk console up, do

set verbose 4

followed by

agi debug

and, if necessary, post the relevant output here. You can turn off agi debugging afterwards with

agi no debug

3- No ring group?

If, by "ring group" you mean making more than one phone ring at once, then the Speed Dials (bad name) section is where you put them. Extension numbers separated by a space.

If you want to make one extension ring and then, if that's busy, another one ring, etc, then that appears to be lacking (Selintra please note!)

Have you had a read of the manual?

http://www.selintra.com/docs/cgi-bin/view/Main/SysVersion2Release1Issue11

[WillKemp]

But i'm pretty sure voicemail should work to some extent.

Checking that out a bit - yeah, voicemail works fine for me. Have you recorded an unavailable message? It should work ok without this, but i've got a feeling it doesn't.

By the way, on the subject of dialing one extension and, if that's busy, dialing a different one, etc, this is the "custom app" i use for that purpose:

Code: [Select]

exten => s,1,Dial(SIP/5000,15)
exten => s,2,GotoIf($["${DIALSTATUS}" = "BUSY"]?3:4)
exten => s,3,Dial(SIP/5001,15)
exten => s,4,Answer
exten => s,n,Voicemail(u5000)
exten => s,n,Hangup

I then select that custom app as the "Open Inbound Route" in the Trunklines config.

[JonB]

Stuntshell,

1 - Voicemail does work and you do not have to record a message for it to work. It will use the default * message (or in my case the Kiwi version of the default message).

Can you dial 300 (conference call) from a phone and get the "you are the only person..." message then music.

If not then I suspect that you do not have a timing source.

Which RPM's did you install?
Did you do a signal-event console-save after installing the SAIL rpm?
Do you have a Zaptel card?
Can you show the output of lsmod?

Jon

[Franco]

@JonB,
Everything else works, I can get into conferences, I can set my voicemail message, on the voicemail the system prompts me the beep as it normally would, but I get no email and if I check my voicemail after (dialing *50*) it tells me I have no messages.
I'm using the X100P card.
Everything was done according to the manual.

Code: [Select]

ztcfg -vv

Zaptel Configuration
======================


Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Ca
1 channels configured.


@WillKemp,
The ring group would ring all phones in that group when a call comes in. In my case I keep one phone in each floor and when someone dials in I can answer where I am. But I like your idea better and will take that.
About the Speed Dial, it seems to do what it says: I can set a extension number to dial a real number or a set of extensions?

About my test with FWD and the question about opening ports:
I created a new carrier for FWD using the IAX2 protocol and that has fixed me as suggested in http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76

But I can only dial out, and not receive:

Code: [Select]

May 28 11:42:11 NOTICE[5766]: chan_iax2.c:6782 socket_read: Rejected connect attempt from 192.246.69.186, who was trying to reach '4613XX@'

[Franco]

Round 2: Reinstalled everything and now I got voicemail  

Still I cannot receive calls from a voip provider  

[SARK devs]

Hello Stuntshell

Thanks for your e-mail re 6.0.  Details on their way shortly.

Still I cannot receive calls from a voip provider

Are you running Server-Gateway or Server-only?  Which VOIP povider are you using and how have you pointed the VOIP Number at your site?  Have you ever received calls from this VOIP provider in the past, perhaps with another asterisk implementation?

Kind Regards

Selintra

[Franco]

Hello and thanks for repplying,
I'll be waiting for the info, thanks!

I'm running Server-Gateway
I'm running FWD and Gizmo for voip providers
In the Trunkline I have set the Open Inbound Route to an internal extension,(not sure if that's what you ask)
Before I had two different setups that worked, SME6 as server-gateway and @home as internal server, and also the same SME6 running dungog's asterisk package.
I do get registered with both providers and I can dial out thru them:
Sip show peers:
174761985XX/174761985XX    198.65.166.131              5060     OK (298 ms)
iax2 show peers:
4613XX/4613XX    192.246.69.186  (S)  255.255.255.255  4569          OK (245 ms)

Firewall on the inbound?
Thanks,