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 Post subject: TRUNK problem
PostPosted: Sat Aug 04, 2007 11:40 am 
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Joined: Mon May 07, 2007 8:27 am
Posts: 38
I just configured A2Billing. When we call our server, it asks for PIN (which is card number) we enter card number, it tells us our balance amount then it asks for destination phone number, we enter phone number, it tells us the minutes to call with available balance. Till here it is working fine, then it tries to place call and do not complete the call.

We have added following entries in SIP.conf

Code:
[mytrunk]
type=peer
host=sip.voip-provider.com
username=myuser
secret=mypassword
context=a2billing


On Asterisk CLI we found the following error:
Code:
Aug  4 11:29:05 WARNING[29380]: channel.c:509 ast_best_codec: Don't know any of 0x0 formats
Aug  4 11:29:05 WARNING[29380]: channel.c:509 ast_best_codec: Don't know any of 0x0 formats
    -- Called mytrunk/009221XXXXXXX
Aug  4 11:29:05 WARNING[29380]: channel.c:2784 ast_channel_make_compatible: No path to translate from SIP/mytrunk-0819f828(0) to SIP/67.15.180.14-081951d8(4)
Aug  4 11:29:05 WARNING[29380]: chan_sip.c:2585 sip_write: Asked to transmit frame type 4, while native formats is 0 (read/write = 0/0)
Aug  4 11:29:05 WARNING[29380]: chan_sip.c:2585 sip_write: Asked to transmit frame type 4, while native formats is 0 (read/write = 0/0)
Aug  4 11:29:05 WARNING[29380]: chan_sip.c:2585 sip_write: Asked to transmit frame type 4, while native formats is 0 (read/write = 0/0)
Aug  4 11:29:05 WARNING[29380]: chan_sip.c:2585 sip_write: Asked to transmit frame type 4, while native formats is 0 (read/write = 0/0)
Aug  4 11:29:05 WARNING[29380]: chan_sip.c:2585 sip_write: Asked to transmit frame type 4, while native formats is 0 (read/write = 0/0)
Aug  4 11:29:05 WARNING[29380]: chan_sip.c:2585 sip_write: Asked to transmit frame type 4, while native formats is 0 (read/write = 0/0)

-- Got SIP response 415 "Unsupported media type"


Then we get response "The number is not responding"
And it asks again to enter phone number.

I would really appreciate if anyone can help in fixing this issue.

Thanks

SK


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 Post subject:
PostPosted: Sat Aug 04, 2007 1:51 pm 
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Joined: Mon May 29, 2006 7:07 pm
Posts: 287
Location: Denver
it's not finding a common codec - add following lines to sip.conf under mytrunk

dissalow=all
allow=ulaw
allow=alaw
allow=gsm


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 Post subject:
PostPosted: Sun Aug 05, 2007 10:01 am 
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Joined: Mon May 07, 2007 8:27 am
Posts: 38
Wow, it worked

Thanks mate.

Well, I could make call and do the normal conversation.

But my credit did not reduce, it may be some configuration issue.

I would really appreciate if anyone can help.

Thanks

SK


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 Post subject:
PostPosted: Tue Aug 07, 2007 11:42 am 
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Joined: Mon May 07, 2007 8:27 am
Posts: 38
Since I was not getting CDR.

I found a solution or work around on this forum.

I downgraded my Asterisk from 1.2.20 to 1.2.18

Now I am getting records and amount is also deducting.

cheers

SK


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