[Asterisk-Users] sipura spa2 + asterisk bug ?

[Tofik Suleymanov]

Hello,

How to reproduce this bug (?) :

1. register sipura spa2 with 2 lines on asterisk.
2. use first line to call somewhere.
3. while using first line try to call from second line somewhere else

in 3 step i hear fast busy tones on second line and asterisk console
gives me this short error:

Mar 27 07:01:43 NOTICE[29656]: chan_sip.c:3629 process_sdp: No
compatible codecs!

My sipura adapter is using g729a codec.
When using both of sipura lines separately everything works fine,
until someone tries to use both lines simultaneously.

Any advice ?


Tofik Suleymanov

[Julian J. M.]

That ATA cannot do 2 simultaneous calls with g729. The second call is
probably trying to use ulaw, alaw or g723. Are you sure any of them
are enabled for that extension?

Julian.

Post by Tofik Suleymanov
Hello,
1. register sipura spa2 with 2 lines on asterisk.
2. use first line to call somewhere.
3. while using first line try to call from second line somewhere else
in 3 step i hear fast busy tones on second line and asterisk console
Mar 27 07:01:43 NOTICE[29656]: chan_sip.c:3629 process_sdp: No
compatible codecs!
My sipura adapter is using g729a codec.
When using both of sipura lines separately everything works fine,
until someone tries to use both lines simultaneously.
Any advice ?
Tofik Suleymanov
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[Steve Kennedy]

Post by Tofik Suleymanov
1. register sipura spa2 with 2 lines on asterisk.
2. use first line to call somewhere.
3. while using first line try to call from second line somewhere else
in 3 step i hear fast busy tones on second line and asterisk console
Mar 27 07:01:43 NOTICE[29656]: chan_sip.c:3629 process_sdp: No
compatible codecs!
My sipura adapter is using g729a codec.
When using both of sipura lines separately everything works fine,
until someone tries to use both lines simultaneously.

The Sipuras only support 1 g.729 codec at any time. If one channel is
using g.729, the other channel has to use another codec.

You'll just have to open more than g.729 at the Asterisk end, so the
first channel can use g.729 and the other another codec.


Steve

[Tofik Suleymanov]

Post by Steve Kennedy

Post by Tofik Suleymanov
1. register sipura spa2 with 2 lines on asterisk.
2. use first line to call somewhere.
3. while using first line try to call from second line somewhere else
in 3 step i hear fast busy tones on second line and asterisk console
Mar 27 07:01:43 NOTICE[29656]: chan_sip.c:3629 process_sdp: No
compatible codecs!
My sipura adapter is using g729a codec.
When using both of sipura lines separately everything works fine,
until someone tries to use both lines simultaneously.

The Sipuras only support 1 g.729 codec at any time. If one channel is
using g.729, the other channel has to use another codec.
You'll just have to open more than g.729 at the Asterisk end, so the
first channel can use g.729 and the other another codec.
Steve

Thanks all for replying
tommorow i'll try to do like you suggested.
One more quick question: why Sipuras cant do more than 1 g.729 channel
at a time ?
Is this somehow related to g.729 licensing ? Is there any other SIP
adapters which override this drawback ?


Tofik Suleymanov

[Vahan Yerkanian]

Post by Tofik Suleymanov
Thanks all for replying
tommorow i'll try to do like you suggested.
One more quick question: why Sipuras cant do more than 1 g.729 channel
at a time ?

Insufficient CPU power to process 2 g729 streams.

Post by Tofik Suleymanov
Is this somehow related to g.729 licensing ? Is there any other SIP
adapters which override this drawback ?

You're looking for Sipura SPA-2100. It has more powerful CPU. Check
http://www.sipura.com/support/spa2100faq/Section_1.html

HTH,
Vahan

[Tofik Suleymanov]

Post by Vahan Yerkanian

Post by Tofik Suleymanov
Thanks all for replying
tommorow i'll try to do like you suggested.
One more quick question: why Sipuras cant do more than 1 g.729 channel
at a time ?

Insufficient CPU power to process 2 g729 streams.

Post by Tofik Suleymanov
Is this somehow related to g.729 licensing ? Is there any other SIP
adapters which override this drawback ?

You're looking for Sipura SPA-2100. It has more powerful CPU. Check
http://www.sipura.com/support/spa2100faq/Section_1.html
HTH,
Vahan
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Vahan, thank you for reply.
i've changed codec types for my sipura lines and now 2 simultaneous
calls from my sipura device to outworld are working, but here is the
other problem i came across:

1. assume 1-st line is in use
2. after dialing 2-nd line from outside i immediately go to the
voicemail announcement (also i immediately go to voicemail if i dial
from extension to extension both of which are on the same sipura device)

Any clues on what is happening ?

Thank you,
Tofik Suleymanov

[Vahan Yerkanian]

Post by Tofik Suleymanov
1. assume 1-st line is in use
2. after dialing 2-nd line from outside i immediately go to the
voicemail announcement (also i immediately go to voicemail if i dial
from extension to extension both of which are on the same sipura device)

Check what response code appears on Asterisk CLI when you dial 2nd line.
If your SPA-2k is SPA-2000 or SPA-2002 there might be a codec
combination that is still overloading CPU and it's sending back
unavailable response. I assume both extensions have separate
username/passwords, don't they?

Vahan

[Tofik Suleymanov]

Post by Vahan Yerkanian

Post by Tofik Suleymanov
1. assume 1-st line is in use
2. after dialing 2-nd line from outside i immediately go to the
voicemail announcement (also i immediately go to voicemail if i dial
from extension to extension both of which are on the same sipura device)

Check what response code appears on Asterisk CLI when you dial 2nd line.
If your SPA-2k is SPA-2000 or SPA-2002 there might be a codec
combination that is still overloading CPU and it's sending back
unavailable response. I assume both extensions have separate
username/passwords, don't they?
Vahan
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Thank you, Vahan
i've managed both of lines to work after playing a bit with codecs.

[Tofik Suleymanov]

Each of the two lines have their own entry in sip.conf and i can see
each line registered in 'sip show peers'.
I can dial each line from outside successfully but when one line is busy
i can't reach the second line (it immediately sends me to the
voicemail).I've also tried to change the timeouts in dial command but
seems that it doesn't matter.
Any other advice ?

You haven't got codec negotiation set-up properly so it's still running
out of g.729 and then it will act as busy
I have
dtmfmode=rfc2833
disallow=all
allow=g729
allow=gsm
allow=alaw
allow=ulaw
allow=g723.1
So should try g.729 first, then gsm (which unfortunately SPA don't
support), etc etc.
Steve

Thank you very much !
after playing a bit with codecs i've managed my sipura lines to work
properly.

Again, thank you very much for quick and effective help !

Tofik Suleymanov