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How to force codec transcoding?

Hi, we are working on a new codec for asterisk. It’s early stages and
the main goal is just to be able to “hear it” using normal client end
points. So what I want is to be able to take a couple of normal G711
extensions, but have asterisk internally force a transcode to my new
codec. In this way we can easily run up a bunch of listening tests with
a wide variety of users very easily

Ok, so how do I force this transcode in the middle, ideally without
using two asterisk boxes?

I clearly need two extensions to have (say) allow=g711, which means I
need to persuade asterisk to route the call via a middle extension which
is set to only allow the new codec. What are my options to achieve
this? (meetme just occurred to me? could that work?) Any suggestions?

Thanks

Ed W

Asterisk 10.3.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 10.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 10.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you!

The following are the issues resolved in this release:

* — Fix potential buffer overrun and memory leak when executing “sip
show peers”
(Closes issue ASTERISK-19231. Reported by Thomas Arimont, Jamuel Starkey)

* — Fix ACK routing for non-2xx responses.
(Closes issue ASTERISK-19389.)

* — Remove possible segfaults from res_odbc by adding locks around
usage of odbc handle
(Closes issue ASTERISK-19011. Reported by Walter Doekes)

* — Fix blind transfer parking issues if the dialed extension is not
recognized as a parking extension.
(Closes issue ASTERISK-19322. Reported by aragon)

* — Copy CDR variables when set during a bridge
(Closes issue ASTERISK-16990.)

* — push ‘outgoing’ flag from sig_XXX up to chan_dahdi
(Closes issue ASTERISK-19316. Reported by Jeremy Pepper)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.3.0

Thank you for your continued support of Asterisk!

 

Asterisk 1.8.11.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.8.11.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 1.8.11.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

* — Fix potential buffer overrun and memory leak when executing “sip
show peers”
(Closes issue ASTERISK-19231. Reported by Thomas Arimont, Jamuel Starkey)

* — Fix ACK routing for non-2xx responses.
(Closes issue ASTERISK-19389.)

* — Remove possible segfaults from res_odbc by adding locks around
usage of odbc handle
(Closes issue ASTERISK-19011. Reported by Walter Doekes)

* — Fix blind transfer parking issues if the dialed extension is not
recognized as a parking extension.
(Closes issue ASTERISK-19322. Reported by aragon)

* — Copy CDR variables when set during a bridge
(Closes issue ASTERISK-16990.)

* — push ‘outgoing’ flag from sig_XXX up to chan_dahdi
(Closes issue ASTERISK-19316. Reported by Jeremy Pepper)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.11.0

Thank you for your continued support of Asterisk!

Mute DTMF

I have been breaking my head on this, can’t find a solution.

Anyone know a way to mute DTMF on SIP? I have already tried changing the
dtmfmode option and messing with different codec/dtmfmode settings but so
far, not having any luck.

Not even sure changing the codec is an option but pulling at straws at the
moment.

Thanks in advance for any assistance.

DAHDI works, but returns CHANUNAVAIL ??

DAHDI 2.6.0, dahdi show status
Description Alarms IRQ bpviol CRC
Fra Codi Options LBO
Wildcard TDM400P REV I Board 5 OK 0 0 0
CAS Unk 0 db (CSU)/0-133 feet (DSX-1)

Dahdi 1 is an internal extension, dahdi 4 is pstn.

This call completes. But DAHDI comes back with CHANUNAVAIL. This a
problem since we then test for CHANUNAVAIL to use an alternative provider.

Types of bridging

Earlier I was using asterisk 1.4 and 1.6. In these version it used to
do native bridging and the CPU load was not very high. Now after
switching to asterisk 1.8 it has started to do remote bridging and the
CPU load has often started to peak.

Could this be a configuration issue. I have done the same SIP settings
that was earlier there in 1.4 and 1.6. I have ‘directmedia=yes’ and
‘directrtpsetup=yes’ in sip.conf and both the peers use the same
codecs and there are no nat issues as well

Please help

On Thu, Mar 29, 2012 at 7:29 PM, Phil Frost wrote:
> On Mar 29, 2012, at 08:43 , Deepesh D wrote:
>> What are the different type of bridging used by asterisk in a SIP
>> call? What is the difference between Packet2Packet bridging, Remote
>> bridging and Native bridging?
>
> Packet2Packet bridging is when RTP datagrams are forwarded by Asterisk without modification. This imposes little load on the CPU. Obviously this can only happen if both ends are using the same codec, and likely there are likely other less obvious conditions that must be met.
>
> Remote bridging happens when Asterisk can direct both ends to send media (RTP probably) to each other directly, by a SIP reINVITE, for example. Only works if both ends have a route to each other, Asterisk is configured to do it, each end shares a codec, and probably a dozen other more subtle conditions are true. In this case there is no load on Asterisk as it’s not even in the media path. It also means it can’t do things like intercept and act on DTMF or monitor the call.
>
> Native bridging is when media is forwarded with Asterisk, but for whatever reason (different codecs, maybe) Asterisk must inspect or modify the stream. Could mean a significant CPU load.
> –
> Phil Frost
> Macprofessionals
> office 248-893-0738
> direct 248-662-0809
>
>
>
>
> –
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