Topic: [Announce] Selintra-sail-2.1.13-256

[marcelb]

Hi Selintra,

The problem is that i have to little understanding of the workings of linux/asterisk/telco stuff. So i'm not able to 'see' the programming trouble behind this and am probably not seeing the bigger picture... but in my laymans thinking:

It's about a logical place where to (be able to) define an outbound caller id.

My first guess would be a outbound caller id field when creating a route, maybe even for every path, that would leave the most flexibility.

Otherwise i'd figure when adding a PTT_DiD trunk also assigning it to a (or more) channel or group. When creating outbout routes beeing able to select a PTT_DiD would hold the information where to place the call and what to use for the Caller ID. But this would probably mean trouble with PTT_DiD number ranges and less flexibility.

Probably my laymans thinking would screw up the entire PBX but anyway... just trying to help the best i can.

Kind regards,
jester.

Selintra / Jester


In version 2.1.13-256 was the option Transformation Mask: in modify Thrunks.
In my situation the NT1 KPN box give me the number 0485 311 xxx as 485 311 xxx.
What i did is make a Transformation Mask: with the following option 485:0485 and oudbound calling worked good. But in this release ther is no Transformation Mask: anymore.

Maybe this will help

Regards Marcel

[del]

Hi Jeff.

Thanks for the help, can you tell me is possible to specify an outgoing VoIP provder (voipbuster for example) and a different one for incoming calls?

Regards,
Del

[SARK devs]

Hi Marcel,

But in this release there is no Transformation Mask: anymore.

The transformation mask is still there.  However you apply it on the base ZAP line not the DiD.

Jester

Does this work for you - if you dial without the leading zero on your ISDN line?

Kind Regards

Selintra

[del]

Hi Jeff,

I am having a few basic problems, I have checked the admin guide but can't seem to find out what I am doing wrong.    I still can't retreive voicemail. it still syas login is incorrect even using the ext # as the password. The other thing I am struggling to get to grips with is making out sides calls, I have added a trunk using voipbuster and put in my user name and password but I can't seem to dial out (I have credit on my account). Can you point me to the instructions in the admin guide please. Thanks again.

Regards,
Del

[SARK devs]

can you tell me is possible to specify an outgoing VoIP provder (voipbuster for example) and a different one for incoming calls?

Hi Del,

Yes it is. You can have as many VOIP providers (and VOIP Numbers) as you wish.  You can also use them for different purposes. Simply add a trunk entry for each carrier you have an account with.  To use a VOIP carrier for outbound you create a Route,  or define a "carrier select" prefix in the trunk.  The trunk decides what happens to inbound calls from a particular carrier.  You can read about routes here

http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter251

You can read about Trunks here

http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter09


Kind Regards

Selintra

[chris burnat]

I still can't retreive voicemail. it still syas login is incorrect even using the ext # as the password.

Check DTMF settings on your phone.  
I have noticed that voicemail cannot be accessed  if INBAND is selected - password fails.
You need INFO, or INFO + INBAND, or AUTO.

[del]

Hi Burnat,

I am using sjphone and will check the dtmf settings, although I have broken my system and I can't even make internal calls anymore. So I am at this moment formatting my machine and will be doing a fresh install. I not sure that I have used the correct rpms as there were so many to choose from!
These were the ones I used:
smeserver-asterisk-zappri-MPP-1.2.2-1.i686.rpm*
smeserver-asterisk-1.2.3-2.i686.rpm
smeserver-sounds-1.2.2-2.i686.rpm
selintra-sail-2.1.13-261.noarch.rpm
Please let me know if I am missing any. Thanks in advance.

Regards,
Del

[SARK devs]

Hi Del,

First of all, apologies for the rpm confusion.  We are rationalising them as we speak to make things less complex.  

The sail rpm -261 (which you are using)  is very much an alpha/beta release at the moment and has known problems which we are currently working on.

For now, I would suggest you remove it (with rpm -e selintra-sail) and instead use 2.1.11-214 which is the current "stable" release.  

Kind Regards

Selintra

[marcelb]

can you tell me is possible to specify an outgoing VoIP provder (voipbuster for example) and a different one for incoming calls?

Hi Del,

Yes it is. You can have as many VOIP providers (and VOIP Numbers) as you wish.  You can also use them for different purposes. Simply add a trunk entry for each carrier you have an account with.  To use a VOIP carrier for outbound you create a Route,  or define a "carrier select" prefix in the trunk.  The trunk decides what happens to inbound calls from a particular carrier.  You can read about routes here

http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter251

You can read about Trunks here

http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter09


Kind Regards

Selintra

Del,

We also use SjPhone with, in my case, sipdiscount.
Please program a carrier with the following Registration Template (Optional):
login:password@sipprovider/phonenumber and it will work.

Regards Marcel

[jester]

Hi Jeff/Selintra,

Nope sorry, i've tried the lot but to no avail. I've put in  _. as the dialplan for my route and tried with leading zero's, without, with int. country code but no luck  (Btw the transformation mask on the ISDNHFC trunk does not yet work).

The weird thing though, is that a message from my operator is played so i'm getting outside. I think i'll try removing it all, remove the db and reinstall the lot to see if that will work.

Kind regards,
jester.

[del]

Hi All,

Thanks for all the help so far.

Jeff, there is a folder on the contribs page called "Non ISDN" it contains different rpms to the ones I have used. As I am only going to be using VoIP not ISDN should I be using these? Thanks again.

Marcel, thanks for the info I will try that once I can make internal calls. At the moment I can dial another ext, it rings but there is no voice transmission either way, funny thing is if I use the SIP PC to PC profile on sjphone I can talk accross the network. I will keep on perservering I am sure that I am probably missing something simple here. I have downloaded the pdf manual from selintra and I will take a little more time to read the basics.

Regards,
Del

[SARK devs]

Jeff, there is a folder on the contribs page called "Non ISDN" it contains different rpms to the ones I have used. As I am only going to be using VoIP not ISDN should I be using these? Thanks again.

Hi Del,

No, those also are experimental (alpha) rpms.  Stay with the ones you have for now and with 2.1.11-214 if you aren't using ISDN.

Possible reasons for no sound...

You are running an internal network other than 192.168.1.X, in which case you need to set "localnet=" (in sip.conf headers) to match your local address range.

You are running server-only and have masq running on the server.  You can check with iptraf to see if the rtp packets are arriving at your server and being dropped by masq.  Easiest is to turn masq off (as long as you have a real firewall running somewhere upstream).  

config setprop masq status disabled

Best to reboot your system after this.  

To turn masq back on...

config setprop masq status enabled

See if that helps.

Kind Regards

Selintra

[del]

Hi Jeff,

Thanks, I somehow broke the server uninstalling the selintra-sail rpm! So I am formatting and reinstalling from scratch  
The IP range could/will be the sound problem as I am using 10.0.0.X range, were will I find the sip.conf headers, can I use pico to edit this? Thanks for the pointers.  

Regards,
Del  

[SARK devs]

were will I find the sip.conf headers, can I use pico to edit this?

Hi Del,

No you won't need pico.  The headers are exposed for you in the headers panel of Sail.  Simply open the panel and click on the sip.conf update icon.  This will show the header tuples in a freeform window and you can set the correct value for your system.

Kind Regards

Selintra

[del]

Hi Jeff,

Thanks, I am actually having trouble installing SME at the moment!
The normal kernel panic stuff, when I am done I will post back with my progress.

Regards,
Del