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chrisgi

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Hello,

I would like to make a new subject to present you the solution we are seting up and the problems we got.
Server is located in France. Phone numbers are 10 digits "XX ** ** ** **" Where XX specifies the type of target :
- 01/02/03/04/05/09 Reaches landlines
- 06/07 Reaches cellphones network
- 08 Reaches special numbers over taxed usually

Emergency numbers are two digits short numbers 15, 17, 18


ISP side : 1 OVH account with two simultaneous call allowed

End user side :
- 1 IP Phone Linksys SPA-922 (Extension 10)
- 1 dual RTC to IP converter Linksys PAP2T (Extension 11 and 12)
- 1 softPhone MacOS X Telephone (Extension 20)
- 1 softPhone Windows Ekiga (Extension 21)

Kerio operator side :
- Provisioning disabled
- 4 users/password/extension + admin account
- Call routing Configuration :
* 1 interface called OVH
* Hostname sip.ovh.net
* Port/User and password are filed with information from operator
* Must register is checked
* User ID differs from telephone number is not checked
* Send the following external ip address in SIP registration is not checked
* Route incoming calls is sent to extension 10. Fallback to extension 12


After all this set up we encounter a couple of problems :

Problem 1 : Interface OVH won't connect. Message : "Interface is not registered or configured, please refresh [...]"

Solution : User ID differs from tepehone number is now checked (but tel number is international form 00339*********, user id is national form 09********)
(so the message #75202 posted on this forum can be closed)


Test : Outgoing call works (Extension 20 to external number)
Test : Incoming call works (External to internal) but the fallback rings (Extension 12)


P 2 : On incoming calls, the fallback number rings, not the route for incoming calls

S : Need help on this


P 3 : I have a log problem, this line is pollution the logs
"[10/Nov/2010 12:10:11] asterisk[1724]: WARNING[1747]: chan_sip.c:6799 in determine_firstline_parts: Bad request protocol Packet"
I got it many times by second. A quick search indicates that it may be the "Registration value in SIP parameter" that should be set to 3600. But I don't know if I can change this parameter

S : Need help on this (already posted on message #75407)

P 4 : Outgoing calls don't work any more on linksys equipements (both). Incoming voice is OK, outgoing is not when call is initiated from the insode. Last change was to move internal mapping and fallback to another phone. No reboot will help.

S : Need help on this.


P 5 : I got this message in Warning log :
[10/Nov/2010 17:12:16] asterisk[1724]: WARNING[6339]: channel.c:3201 in ast_channel_make_compatible: No path to translate from SIP/10-08c64908(Cool to SIP/0033972127415-08c63378(1024)
[10/Nov/2010 17:12:17] asterisk[1724]: WARNING[6339]: channel.c:2811 in set_format: Unable to find a codec translation path from ilbc to slin
[10/Nov/2010 17:12:17] asterisk[1724]: WARNING[6339]: file.c:912 in ast_streamfile: Unable to open vm-intro (format 0x400 (ilbc)): No such file or directory

S : Single reboot does not change anything. Sorry about the very bad diagnostic way Wink
S2 : The OVH link was on "GSM" (0x1e0e) codec. I took it out so the link went down and back up with "ulaw" (0x1e0c) codec. Test call = Not OK
S3 : The phone (Linksys SPA-922) was on G723 Codec, I switched it to G711u. Test call = Not OK


P 6 : On the admin interface, back and forward does not disconnect the user

S : Need help on this.


P 7 : Scroling back in logs often crashes the log interface

S : Need help on this.


P 8 : Busy and away messages recorded through the audio interface are not published. The messages stay in original english.

S : Need help on this.


P 9 : Calls on hold are not notified by any voice message or holding beep

S : Need help on this.


So, what do you think about this outages ?

Thanks for all folks and good luck.

[Updated on: Sat, 13 November 2010 01:15]

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Vladimir Toncar (Kerio)

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Hi,

First of all, please check whether your trial is still valid. If not, I will send you a license file.

Vladimir
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chrisgi

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Yes it is still valid for the moment.

But u can send us a better one Wink

[Updated on: Mon, 15 November 2010 15:55]

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Vladimir Toncar (Kerio)

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Hi,

I can confirm that P2 is a bug. We were able to reproduce this behavior. Will keep you updated about the rest.

Vladimir
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chrisgi

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An update on our testings. I discovered something that I forgot to mention in my previos testings :

P10 :
Step 1 : change codecs inthe SIP interface for external provider connexion (deleteting or moving) codecs.
Step 2 : Make a change on Extension/Internal maping or fallback number and applie the changes
Then the codec of the interface are back to the original list/order

Thx for your help
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chrisgi

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Good evening,
If you follow the link below, you will find the procedure from our SIP provider to interconnect Asterisk with them.

http://guides.ovh.com/pdf/fr/AsteriskEtForfaitOVH_14.01.2010 .14:53.pdf

Regards
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