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Selintra + sme7.3+ Freshtel VSP

Offline groutley

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« on: June 30, 2007, 02:03:05 AM »
Hello,
 I am hoping for some help,
I am a newbie to Asterisk and the Selintra Contrib is my first attempt.

I have extensions working using x-Lite, and can call xtn to xtn, and use voicemail.
I have defined one Trunk with Pennytel, and this works fine. I can dial a landline from an X-Lite Extension !!

I am also trying to define my Freshtel accounts as Trunks, but I have no joy.
I have tried with the predefined Freshtel SIP carrier, I have also tried to create an IAX2 Freshtel carrier and use that.
But no matter what I do, the Trunk Status is a Red X and clicking on it tells me it cannot find the Peer. 'Peer FreshtelIAX2 not found.'

now my searches find various howtos on Freshtel with Asterisk, but I am not certain how to combine these howtos with using SAIL to configure.
e.g.
http://www.voip-info.org/wiki/view/Asterisk+with+Freshtel
my main concern is that the howtos seem to suggest that the 'context' needs to be changed,  but from what I gather with the SAIL integration this should always be 'mainmenu'.

Here is the asterisk output when I commit my Freshtel IAX2 trunkline...

[Jun 30 09:29:27] NOTICE[26551]: cdr.c:1093 do_reload: CDR simple logging enabled.
[Jun 30 09:29:27] NOTICE[26551]: indications.c:505 ast_unregister_indication_country: Removed default indication country 'au'
[Jun 30 09:29:27] WARNING[26551]: res_smdi.c:746 reload: No SMDI interfaces were specified to listen on, not starting SDMI listener.
[Jun 30 09:29:27] WARNING[26551]: chan_misdn.c:991 reload_config: chan_misdn is not initialized properly, still reloading ?
[Jun 30 09:29:27] NOTICE[26551]: app_playback.c:455 reload: Reloading say.conf
[Jun 30 09:29:27] WARNING[26551]: pbx.c:6231 ast_context_verify_includes: Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls'
[Jun 30 09:29:27] WARNING[26551]: pbx.c:6231 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-canonical'
[Jun 30 09:29:27] WARNING[26551]: pbx.c:6231 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-customers'
[Jun 30 09:29:27] WARNING[26551]: pbx.c:6231 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-via-pstn'
[Jun 30 09:29:27] WARNING[4372]: config.c:775 process_text_line: No '=' (equal sign) in line 98 of /etc/asterisk/sip.conf
[Jun 30 09:29:27] WARNING[4372]: config.c:775 process_text_line: No '=' (equal sign) in line 99 of /etc/asterisk/sip.conf
[Jun 30 09:29:27] WARNING[4372]: chan_sip.c:16393 reload_config: tos value at line 11 is deprecated.  See doc/ip-tos.txt for more information.
[Jun 30 09:29:27] NOTICE[26551]: pbx_ael.c:3910 pbx_load_module: Starting AEL load process.
[Jun 30 09:29:27] NOTICE[26551]: pbx_ael.c:3917 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'.
[Jun 30 09:29:27] WARNING[26551]: ael.y:205 ael_yyparse: ==== File: /etc/asterisk/extensions.ael, Line 112, Cols: 34-34: Warning! The empty context ael-dundi-e164-canonical will be IGNORED!
[Jun 30 09:29:27] WARNING[26551]: ael.y:205 ael_yyparse: ==== File: /etc/asterisk/extensions.ael, Line 120, Cols: 34-34: Warning! The empty context ael-dundi-e164-customers will be IGNORED!
[Jun 30 09:29:27] WARNING[26551]: ael.y:205 ael_yyparse: ==== File: /etc/asterisk/extensions.ael, Line 128, Cols: 33-33: Warning! The empty context ael-dundi-e164-via-pstn will be IGNORED!
[Jun 30 09:29:27] NOTICE[26551]: pbx_ael.c:3925 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'.
[Jun 30 09:29:27] WARNING[26551]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-canonical' cannot be found.
[Jun 30 09:29:27] WARNING[26551]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-customers' cannot be found.
[Jun 30 09:29:27] WARNING[26551]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-via-pstn' cannot be found.
[Jun 30 09:29:27] WARNING[26551]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 276-283: The included context 'ael-parkedcalls' cannot be found.
[Jun 30 09:29:27] NOTICE[26551]: pbx_ael.c:3928 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'.
[Jun 30 09:29:27] NOTICE[26551]: pbx_ael.c:3930 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'.
[Jun 30 09:29:27] NOTICE[26551]: pbx_ael.c:3933 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'.
[Jun 30 09:29:27] WARNING[26551]: pbx.c:6231 ast_context_verify_includes: Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls'
[Jun 30 09:29:27] WARNING[26551]: pbx.c:6231 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-canonical'
[Jun 30 09:29:27] WARNING[26551]: pbx.c:6231 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-customers'
[Jun 30 09:29:27] WARNING[26551]: pbx.c:6231 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-via-pstn'
[Jun 30 09:29:27] NOTICE[26551]: pbx_ael.c:3936 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'.
l1nuxsvr*CLI>

Any help would really be appreciated..
  Glen

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« Reply #1 on: June 30, 2007, 11:23:31 AM »
Hi Glen,

We're a little confused so if we could ask a few questions...

You say you attempted to use the pre-defined Freshtel SIP definition.  Er... we didn't know we had one.  It isn't in the distribution database.  So... which definition did you use?  

We are assuming you have an account set up with Freshtel...  Yes?  If so, does it have an inbound DiD (telephone number) associated with it? (we don't need to know what the number is, just whether you have been allocated one or not).

Finally, is it a SIP account or an IAX account?

Kind Regards

Selintra

Offline groutley

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« Reply #2 on: June 30, 2007, 11:47:49 AM »
Quote from: "selintra"

You say you attempted to use the pre-defined Freshtel SIP definition.  Er... we didn't know we had one.  It isn't in the distribution database.  So... which definition did you use?  

Oooops,  sorry  I wondered if I added that one as well, just been playing with this for a while now, I forgot what was and wasn't there.

Quote from: "selintra"

We are assuming you have an account set up with Freshtel...  Yes?  If so, does it have an inbound DiD (telephone number) associated with it? (we don't need to know what the number is, just whether you have been allocated one or not).

Yes, I have a number of freshtel accounts

Quote from: "selintra"

Finally, is it a SIP account or an IAX account?


Frestel accounts can be accessed by either IAX or SIP depending on the gateway used.
gateway.freshtel.net  for IAX
sip.freshtel.net  for SIP

So I can use any account as either SIP or IAX I just need to point it to the respective gateway.

again, sorry for the confusion I caused..

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« Reply #3 on: June 30, 2007, 03:19:34 PM »
Hello Glen,

We have tested both IAX2 and SIP connections with Freshtel.  To follow our instructions please first read...

http://selintra.com/docs/cgi-bin/view/Main/DocChapter259

Delete your existing Freshtel carrier definition(s).  
Commit.  
Now, create a new Freshtel carrier (SIP first)  EXACTLY as we do for Telasip in the above document (include a regsitration string with the correct Freshtel IP address).  NB, the registration string accepts the substitutional variables username and password.  You enter them as "username:password" and sail will substitute the real entries at trunk creation time.  Add insecure=very somewhere in the peer entry (just as we do in the example).    

Commit your new carrier.

Create a new Trunk and select your Freshtel carrier from the dropdown.
Fill out the fields as follows:-

In the DiD number field enter your 8-digit Freshtel/Firefly number
In the Peer stanza label put a unique string (e.g. ftel1,  or something similar)
In the Username field enter your 8-digit Freshtel/Firefly number (again)
In the Password field enter your Freshtel password for this account.

Commit.

That's it, you're done.  Your trunk panel should now show your Freshtel account with a nice "connected" icon.

The set-up for IAX2 and SIP is identical except for the freshtel url.

If you have any Freshtel DiDs which come into this account then simply create one additional PTT-DiD trunk for each DiD.

Kind Regards

Selintra

Offline groutley

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« Reply #4 on: June 30, 2007, 11:58:48 PM »
Thanks Selintra,
  I guess I messed up the SIP carrier definition, and then assumed it couldn't be wrong as I thought it was provided.. Doh !

So following your instructions for the SIP it worked !!  whoohoo..  Thanks !

Although I still have no luck on the IAX carrier..
After success with the SIP carrier, I then followed the same process for IAX2
only defining the URI different..
 I then created the Trunkline from this IAX2 carrier def.
and I get the same problem as before..  cant find peer..
here is the asterisk log when I commit the IAX trunkline....
Code: [Select]
[Jul  1 07:40:36] NOTICE[6621]: cdr.c:1093 do_reload: CDR simple logging enabled.
[Jul  1 07:40:36] NOTICE[6621]: indications.c:505 ast_unregister_indication_country: Removed default indication country 'au'
[Jul  1 07:40:36] WARNING[6621]: res_smdi.c:746 reload: No SMDI interfaces were specified to listen on, not starting SDMI listener.
[Jul  1 07:40:36] WARNING[6621]: chan_misdn.c:991 reload_config: chan_misdn is not initialized properly, still reloading ?
[Jul  1 07:40:36] NOTICE[6621]: app_playback.c:455 reload: Reloading say.conf
[Jul  1 07:40:36] WARNING[6621]: pbx.c:6231 ast_context_verify_includes: Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls'
[Jul  1 07:40:36] WARNING[6621]: pbx.c:6231 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-canonical'
[Jul  1 07:40:36] WARNING[6621]: pbx.c:6231 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-customers'
[Jul  1 07:40:36] WARNING[6621]: pbx.c:6231 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-via-pstn'
[Jul  1 07:40:36] NOTICE[6621]: pbx_ael.c:3910 pbx_load_module: Starting AEL load process.
[Jul  1 07:40:36] NOTICE[6621]: pbx_ael.c:3917 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'.
[Jul  1 07:40:36] WARNING[6621]: ael.y:205 ael_yyparse: ==== File: /etc/asterisk/extensions.ael, Line 112, Cols: 34-34: Warning! The empty context ael-dundi-e164-canonical will be IGNORED!
[Jul  1 07:40:36] WARNING[6621]: ael.y:205 ael_yyparse: ==== File: /etc/asterisk/extensions.ael, Line 120, Cols: 34-34: Warning! The empty context ael-dundi-e164-customers will be IGNORED!
[Jul  1 07:40:36] WARNING[6621]: ael.y:205 ael_yyparse: ==== File: /etc/asterisk/extensions.ael, Line 128, Cols: 33-33: Warning! The empty context ael-dundi-e164-via-pstn will be IGNORED!
[Jul  1 07:40:36] NOTICE[6621]: pbx_ael.c:3925 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'.
[Jul  1 07:40:36] WARNING[6621]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-canonical' cannot be found.
[Jul  1 07:40:36] WARNING[6621]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-customers' cannot be found.
[Jul  1 07:40:36] WARNING[6621]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-via-pstn' cannot be found.
[Jul  1 07:40:36] WARNING[6621]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 276-283: The included context 'ael-parkedcalls' cannot be found.
[Jul  1 07:40:36] NOTICE[6621]: pbx_ael.c:3928 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'.
[Jul  1 07:40:36] WARNING[4753]: config.c:775 process_text_line: No '=' (equal sign) in line 98 of /etc/asterisk/sip.conf
[Jul  1 07:40:36] WARNING[4753]: config.c:775 process_text_line: No '=' (equal sign) in line 99 of /etc/asterisk/sip.conf
[Jul  1 07:40:36] NOTICE[6621]: pbx_ael.c:3930 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'.
[Jul  1 07:40:36] NOTICE[6621]: pbx_ael.c:3933 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'.
[Jul  1 07:40:36] WARNING[6621]: pbx.c:6231 ast_context_verify_includes: Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls'
[Jul  1 07:40:36] WARNING[6621]: pbx.c:6231 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-canonical'
[Jul  1 07:40:36] WARNING[6621]: pbx.c:6231 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-customers'
[Jul  1 07:40:36] WARNING[6621]: pbx.c:6231 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-via-pstn'
[Jul  1 07:40:36] NOTICE[6621]: pbx_ael.c:3936 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'.
[Jul  1 07:40:36] WARNING[4753]: chan_sip.c:16393 reload_config: tos value at line 11 is deprecated.  See doc/ip-tos.txt for more information.
l1nuxsvr*CLI>

Also, I did find the Template detail a little confusing,  the Documentation URL unfortunately doesn't display what exactly was entered in the field.
Also I cut and pasted from that document page "user-id:password@carrierURI/deliverytag" and edited the carrierURI,
I was wondering if the username:password are substituted, why the carrierURI and delivery tag are not..
I did try !!
Also then noted as quoted in the Document page "user-id:" is incorrect and should be username.
so my template ended up "username:password@sip.freshtel.net/username"
This worked nicely :-)
I would like to understand the IAX issue..  although I can get by with SIP, the IAX with freshtel gives an online indicator to others on the Firefly softphone, this would be very handy for my kids.

Thank you very much for the help so far..
 Glen

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« Reply #5 on: July 01, 2007, 11:50:35 AM »
Hi Glen,

Glad the SIP def worked.  Our IAX version is fine too...

Here are the actual IAX.conf fragments we ended up with for the Trunk...

peer entry
Code: [Select]
type=peer
host=gateway.freshtel.net
qualify=3000
canreinvite=no
username=84972704
fromuser=84972704
secret=xxxxxxx
disallow=all
allow=alaw
allow=ulaw


user entry
Code: [Select]
type=user
context=mainmenu


This gives the correct "connected" icon in the trunks panel, although, due to a small bug we've just found, if you click on the icon it will say "peer not found"- this will be fixed in next release of the code.

Kind Regards

Selintra[/code]

Offline groutley

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« Reply #6 on: July 01, 2007, 01:02:48 PM »
Hi Selintra,
  very curious..  here is my trunk section of IAX.conf
Code: [Select]
;External IAX2 lines
;
;==========================================================================
;      This is where we deal with IAX2 trunks (if any).
;      We generate 2 entries (peer and user) for each.
;==========================================================================
;

;

Nothing there !!
Yet looking in the SAIL trunkline def  its is all there..
Code: [Select]
ftel0112
type=peer
host=gateway.freshtel.net
qualify=3000
canreinvite=no
username=87950112
fromuser=87950112
secret=xxxxx
disallow=all
allow=alaw
allow=ulaw
87950112
type=user
context=mainmenu


I 're'committed after copying the above..
Operation status report - Updates have been committed
and iax.conf now shows



Code: [Select]
;External IAX2 lines
;
;==========================================================================
;      This is where we deal with IAX2 trunks (if any).
;      We generate 2 entries (peer and user) for each.
;==========================================================================
;

[ftel0112]
type=peer
host=gateway.freshtel.net
qualify=3000
canreinvite=no
username=87950112
fromuser=87950112
secret=xxxxx
disallow=all
allow=alaw
allow=ulaw
[87950112]
type=user
context=mainmenu

;

and up near the top of iax.conf

But still no change to the status..  red X
Code: [Select]
register => 87950112:xxxxx@gateway.freshtel.net/87950112


Although now in Asterisk console
Code: [Select]
l1nuxsvr*CLI> iax2 show peers
Name/Username    Host                 Mask             Port          Status
ftel0112/879501  202.86.48.134   (S)  255.255.255.255  4569          OK (29 ms)

So now I am very confused...  I Know I had definitely committed the changes previously.
And I get a red X on the SAIL display yet CLI suggests otherwise ?
I have not worked out a dialplan yet to test this trunk to verify its operation

Offline groutley

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« Reply #7 on: July 01, 2007, 01:15:35 PM »
Update..
  I noticed in the iax2 show peers as I was posting above that the account number was truncated in the display.
I assumed this would not make any difference and was just a display issue, but I am finatical, and like things to LOOK tidy..
So I deleted the Trunk, and redifined with a shorter Stanza.. 'ft0112'
BINGO !!
a nice connected symbol ! although I do have the bug you mention of not found in the detail..
Do  is the stanza limited to 6 characters ?  and should this be Noted in the documentation?

Now to work out the dial plan..
One question on dial plan I have not worked out, is
In the State that I live in, all numbers are prefixed with '613' and then an 8 digit number is dialed,
With my SPA3102 dial plan I can tell it, if an 8 digit number is dialed to prefix that with a 613 (61 = Australia, 3 = Victoria)
But I cannot find anything on how to accomplish this for asterisk.

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« Reply #8 on: July 02, 2007, 10:31:29 AM »
Hi Groutley

Try reading this link here >> http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter096#Transformation_mask

I did have one setup on my old Faktortel account, but haven't needed it with our current provider and I deleted the Faktortel account through sheer frustration, so can't tell you how it was.

Basically in Vic you would have 8:6138 which will transform any number starting with an 8 to start with 6138 ie; 89450000 would change to 61389450000.

You need to add one for each transform you want there.

Hope that helps.

Mark

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« Reply #9 on: July 02, 2007, 10:56:18 AM »
Hi Selintra,
  I have got that far, and can succeed in dialing Aussie numbers,
I am stuck though at ow to differentiate between a Vic number begining with 8 and dialing 6138  and a Freshtel number which begins with 8 and should not be prefixed.

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« Reply #10 on: July 03, 2007, 12:41:40 AM »
mmmm, didn't think of that one.

I'll have to have a bit of a think about that one, you can probably smell the smoke as the gears begin to turn.

I'll get back to you if I figure a way around it b4 jeff.

Offline groutley

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« Reply #11 on: July 07, 2007, 02:31:18 AM »
Hi again,
  I still cannot work out this dial VSP dial out,
I have tried appending a prefix to the Trunk to use a free VOIP only account, and the prefix still does not get recognised, and goes out another trunk which I gather is substituting the 613, as the VSP suggests I dont have enough credit to make the call.  yet this shoul dbe a free call.

I have also played with provisioning my SPA-3102, which 'kind of' works.
the issue I have is the Line 1 dial plan (*x.|*xx*x.|x.)  gets sent to the SPA as  (*x.¬*xx*x.¬x.)  the SPA does not like the '¬' If I manually overtype the symbol, with '|' the dial plan works.
SO I have disabled the Auto provisioning on the SPA to get around this, but I would like to understand why the character substitution, I assume different code page setting on the SME server being I am in Australia ?

Also with the SPA-3102  I originally used the supplied SPA-3000 device template.  but noticed the TFTP provisioning request was asking for SPA3102.cfg which did not exist,  I tried renaming this in the provisioning field, but the change never takes..
So I created my own SPA-3102 device, but I dont understand something.
the SPA-3000 is listed ONCE in devices,  yet there is a SPA-3000FXO  and an  SPA-3000FXS option depending whether I am adding a Trunk or Extension,  with my home brew SPA-3102 I do not get BOTH ? I created
a SPA-3102FXS which works great on the Extensions,  but I have nothing customised for FXO on the Trunk and I cant see how ?

Other than that I did succeed in creating a Cisco 7940 Phone device and successfully provisioning it !!  Whooohooo..

I have now moved the house over to this Asterisk/SAIL server, and would really like to fix the above problems before the wife actually notices..

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« Reply #12 on: July 07, 2007, 03:02:29 AM »
Spoke too soon, about wife noticing !
 she just tried to dial a 1902XXXXXX number which I had forgotten in any dial plan.
now these numbers must go via PSTN (so my SPA-3102 trunk)
so I create a route to direct them to that trunk.
but she gets a 'sorry you could not be connected' voice message

I notice in the asterisk CLI..
Code: [Select]
[Jul  7 10:51:06] WARNING[4783]: chan_sip.c:11823 handle_response_invite: Received response: "Forbidden" from '"SPA" <sip:0397951738)@192.168.37.251>;tag=as058116f0'
[Jul  7 10:51:07] WARNING[6619]: file.c:553 ast_openstream_full: File call-cannot-complete does not exist in any format
[Jul  7 10:51:07] WARNING[6619]: file.c:804 ast_streamfile: Unable to open call-cannot-complete (format 0x4 (ulaw)): No such file or directory
[Jul  7 10:51:07] WARNING[6619]: pbx.c:5668 pbx_builtin_background: ast_streamfile failed on SIP/5000-084b67a8 for call-cannot-complete

Makes no sense to me !!
I have worked around this by putting 1902 via gw0  on my SPA dialplan,
but I obviously have missed something in Asterisk setup..
While my work around works for the main Home Phone Extension,  o other extension will dial these 1902 numbers .

WAF -1 point

Offline SARK devs

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« Reply #13 on: July 08, 2007, 10:55:24 PM »
Well,

...you have been in the wars!

OK one thing at a time.  The messages you got on the 1902 dial are easily explained.  The sp rejected the call, probably because of the dial plan it runs.  The rest of the messages are benign.  SAIL uses a UK phrase in the english language version which isn't present in any other language.  The error messages are merely telling you that asterisk couldn't fine the phrase which the AGI asked it to play.  No big deal...  :-)

As to Vic numbers and Freshtel both beginning with an 8.  If there is no obvious differentiation between the two number classes (maybe the second or third digits differ?) then you can introduce a difference of your own.  Simply add a prefix to one of the classes (preferably the one you dial the least).  Let's say, for sake of argument that we will dial 9 for Freshtel.  So now you can easily handle the two classes with a mask of..  
Code: [Select]
8:6138 9:
Means you have to dial an extra digit for Freshtel but that shouldn't cause too much inconvenience.

Lastly your 3102.  We'll do a deal with you...  Publish your Cisco provisioning so we can incorporate it into SAIL and we'll fix the 3102 for you.  How does that sound?

Kind Regards

Selintra

Offline groutley

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« Reply #14 on: July 10, 2007, 07:43:43 AM »
Quote from: "selintra"
The messages you got on the 1902 dial are easily explained.  The sp rejected the call, probably because of the dial plan it runs.  The rest of the messages are benign.  SAIL uses a UK phrase in the english language version which isn't present in any other language.  The error messages are merely telling you that asterisk couldn't fine the phrase which the AGI asked it to play.  No big deal...  :-)

OK, that explains the messages, but the point really is that the call doesnt succeed out the PSTN Trunk, In fact further testing shows I cannot route any call via Asterisk out the SPA3102 PSTN port.

Quote
As to Vic numbers and Freshtel both beginning with an 8.  If there is no obvious differentiation between the two number classes (maybe the second or third digits differ?) then you can introduce a difference of your own.  Simply add a prefix to one of the classes (preferably the one you dial the least).  Let's say, for sake of argument that we will dial 9 for Freshtel.  So now you can easily handle the two classes with a mask of..  
Code: [Select]
8:6138 9:
Means you have to dial an extra digit for Freshtel but that shouldn't cause too much inconvenience.

That was the solution I had got to, I dial '87' on a Freshtel number,  my big problem was getting my little brain around the order of Dialing Rules in a Route, and the transformation mask in a Trunk,  once I finally got that clear and managed to send the number to the trunk so that is can be transformed I actually managed to make it work..  last little stumbling block was Freshtel expect the Freshtel number to be prefixed with an '*', so Transformation mask of 878:*8  works a beaut !

Quote

Lastly your 3102.  We'll do a deal with you...  Publish your Cisco provisioning so we can incorporate it into SAIL and we'll fix the 3102 for you.  How does that sound?

Sounds too good to be true !  Happy to publish What I did, while still needing perfecting, i.e.  SAIL does not create the provisioning configs, I manually built them and placed in the tftproot directory. but it works !
Where would you like me to document this ?
If you can solve this 3102 issue for me I'm happy to do anything !