SAIL Community Supported PBX

Ok, I have managed to work this out, after a lot of work! Basically Enta put the DDI number in a custom header called X-DNQ (don't ask me why and the current tech manager doesn't know either) but it is.

So I  was able to add the following to sark_customer_extensions_globals.conf:

[from-custom]
exten => _.,1,Goto(mainmenu,${SIP_HEADER(X-DNQ)},1)

In case anyone happens to come across the same issue!

I actually set the default route to hang up, so it ignores the default route and only sees it based on the main inbound number for the trunk not the ddi that has been called in.

I assume the decision not to add the users call recordings to the user login area was down to not competing with Provu's own Advanced call recording addon? It would have been really nice (and I assume pretty simple) to just add another tab next to the voicemail so that they can listen and download call recordings for their extension.

Below is the output of the SIP debug. So I called from my mobile to DDI ending in 34 but it seems to ignore the route for 34 and goes to the default route.

Any thoughts?

Thanks

Richard

<--- SIP read from UDP:87.127.240.101:5060 --->
ACK sip:441305*****3@52.31.56.45:5060 SIP/2.0
Via: SIP/2.0/UDP 87.127.240.101;branch=z9hG4bKebf1.de7796a4.0
From: "0**********9" <sip:0********9@proxy.entacall.com>;tag=as766a7e24
Call-ID: 5bf7d9021ff873e6505eba747f290c17@proxy.entacall.com
To: <sip:4413052******4@proxy.entacall.com>;tag=as74a912bd
CSeq: 102 ACK
User-Agent: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '5bf7d9021ff873e6505eba747f290c17@proxy.entacall.com' Method: ACK

<--- SIP read from UDP:enduserip:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 52.31.56.45:5060;branch=z9hG4bK693afe5e;rport=5060
From: <sip:405@mysipserver:5060>;tag=as73a526d3
To: "Emma" <sip:404@mysipserver:5060>;tag=3574087852
Call-ID: 0_3898411430@192.168.1.102
CSeq: 268 NOTIFY
User-Agent: Yealink SIP-T42G 29.80.23.1
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:phoneipaddress:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 52.31.56.45:5060;branch=z9hG4bK5dcef61e;rport=5060
From: "0*********9" <sip:0**********9@52.31.56.45>;tag=as02192c47
To: <sip:405@enduserip:5060>;tag=2210439667
Call-ID: 6da5ff4510b6293b293bad4c5d6dbeb5@52.31.56.45:5060
CSeq: 102 CANCEL
User-Agent: Yealink SIP-T42G 29.80.23.1
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:enduserip:5060 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 52.31.56.45:5060;branch=z9hG4bK5dcef61e;rport=5060
From: "0*********9" <sip:0*********9@52.31.56.45>;tag=as02192c47
To: <sip:405@enduserip:5060>;tag=2210439667
Call-ID: 6da5ff4510b6293b293bad4c5d6dbeb5@52.31.56.45:5060
CSeq: 102 INVITE
User-Agent: Yealink SIP-T42G 29.80.23.1
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:405@enduserip:5060> for address/port to send to
set_destination: set destination to enduserip:5060
Transmitting (NAT) to enduserip:5060:
ACK sip:405@enduserip:5060 SIP/2.0
Via: SIP/2.0/UDP 52.31.56.45:5060;branch=z9hG4bK5dcef61e;rport
Max-Forwards: 70
From: "0**********9" <sip:0**********9@52.31.56.45>;tag=as02192c47
To: <sip:405@enduserip:5060>;tag=2210439667
Contact: <sip:0*********9@52.31.56.45:5060>
Call-ID: 6da5ff4510b6293b293bad4c5d6dbeb5@52.31.56.45:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
Content-Length: 0


---
Scheduling destruction of SIP dialog '6da5ff4510b6293b293bad4c5d6dbeb5@52.31.56.45:5060' in 6400 ms (Method: INVITE)

How is the progress on testing with v5, any idea on timescales from Beta to Stable release?

Another weird one, I have just started testing Entacall/Entanet basic SIP trunk service and I'm having major issues getting DDIs being recognised.

I haven't had this problem with any other provider, but basically the SIP Header contains the DDI number that has been dialled and even though I have a route set up for it, asterisk ignores it and goes to the default route.

I have copied the main content from the SIP header

Message Header
    Record-Route: <sip:87.127.240.101;lr;ftag=as70f6c22d>
    Via: SIP/2.0/UDP 87.127.240.101;branch=z9hG4bK98f8.7fb25d16.0
        Transport: UDP
        Sent-by Address: 87.127.240.101
        Branch: z9hG4bK98f8.7fb25d16.0
    Via: SIP/2.0/UDP 87.127.215.242:5060;branch=z9hG4bK3b8a8a3a;rport=5060
        Transport: UDP
        Sent-by Address: 87.127.215.242
        Sent-by port: 5060
        Branch: z9hG4bK3b8a8a3a
        RPort: 5060
    Max-Forwards: 69
    From: "07557********" <sip:07557*******@proxy.entacall.com>;tag=as70f6c22d
        SIP Display info: "07557********"
        SIP from address: sip:07557*******@proxy.entacall.com
        SIP from tag: as70f6c22d
    To: <sip:441308********@proxy.entacall.com>
    Contact: <sip:07557*******@87.127.215.242:5060>
    Call-ID: 3d588bd8288eeb3c6520190d212e3303@proxy.entacall.com
    CSeq: 102 INVITE
    User-Agent: Entanet Media Server
    Date: Tue, 05 Jul 2016 15:45:36 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces
    X-DNQ: 441308********
    Content-Type: application/sdp
    Content-Length: 393

Any thoughts would be appreciated?

Thanks

Richard

Thanks, will take a look at that tomorrow. This will make my customer very happy indeed :-)

Hi

Looking for some assistance, I have a setup that needs to use a different call group depending on the day of the week they are closed i.e.

Monday - Open 09:00 - Closed 15:30 (Open call group 1, closed call group 2)
Tuesday - Open 09:00 - Closed 15:30 (Open call group 1, closed call group 3)
Wednesday - Open 09:00 - Closed 15:30 (Open call group 1, closed call group 2)
Etc, etc.

Any suggestions?

Thanks

Richard

Hi

Just spun up an instance with the new v5 and all seemed to go ok but when I try to dial just the voicemail the logs show the following error 

Launched AGI Script /usr/share/asterisk/agi-bin/sarkhpe
sarkhpe,OutCos,*50*,,: Failed to execute '/usr/share/asterisk/agi-bin/sarkhpe': No such file or directory

I have checked and it's there so I'm assuming it's a permissions issue but not sure what username or group sark is expecting to be using. This is on AWS with Debian Jesse so I know permissions can be a bit funny. I did full install as root though.

Thanks,

Rich

Thanks, I will take a look. Any clues on release dates for the v5 beta?

Hi

Has anyone managed to install on Ubuntu 14.04 LTS, Debian Wheezy isn't supported in AWS and I know that Sail doesn't work on Jesse so just wondered if it's possible to do it and if so, is there much difference in the install method?

Thanks

Richard

Any update on the release of v5 beta? I assume this will work on Debian Jesse as v4 didn't seem to work on it.

Looking forward to testing it out.

I'm just playing with the multi-tenant option at the moment, is there any way of either increasing the length of the extension beyond 4 or alternatively is there any way of setting the 1st digit to be assigned to a specific tenant. i.e. 1401 is tenant 1, 2401 is tenant 2 etc.

Thanks

R

Hi

I am currently using Sail on AWS as a hosted solution and at the moment I am just locking down the firewall to selected IP addresses for handset use, which is OK for single office deployments but I want the system to be more flexible. I wondered about managing the handset security using OpenVPN instead as I know that a number of the Yealink handsets I use now support this but not confident at setting this up.

I wondered if anyone has already done it and could provide me with a bit of a walk through setting it up on sail?

Thanks,

Richard

UPDATE - After realising that creating the image it had actually messed up the original instance as well. I did a seperate fresh instance and found that for some reason when it creates the image it changes file permissions. Set them back to what they should be and all working fine.

Hi

I have been testing for a while a small PBX setup on AWS with the aim of creating my own AMI to make it easier to setup future installs. However I have just tried creating a new instance with the AMI I created and everything appears OK until I try to change the config and it give me the following error SQLSTATE[HY000]: General error: 8 attempt to write a readonly database

I have checked the permission for /opt/sark/db and all files etc seem to have the same permissions as the instance that is running without issue.

Any thoughts on what else I could try?

Thanks,

Richard