No path to translate from SIP

Asterisk Asterisk Support
1

Hi,

I’m trying to setup new asterisk server, currently I’ve configured for outbound but I don’t know why but call don’t go through

I’ve even disabled G729 thinking may be some license issue.

receiving following error:

channel.c:6217 ast_channel_make_compatible_helper: No path to translate from SIP/voxbeam-00000028 to SIP/800-00000027
== Spawn extension (outbound1, 19092458350, 1) exited non-zero on ‘SIP/800-00000027’

2

This is codec issue , this thread will give you a more detailed explanation http://forums.asterisk.org/viewtopic.php?f=1&t=80074

3

Ok…but how to fix this issue?

I tried core show translation paths g729 but there was no translation path.

4

I’d suggest providing the version of Asterisk in use, the configuration for sip.conf, and the complete console output with “sip set debug on” done. It will show what is actually being negotiated.

5

I’d also add the output of “core show translations”.

6

Asterisk version: 11.25.1

output of core show translations:

Translation times between formats (in microseconds) for one second of data
          Source Format (Rows) Destination Format (Columns)

            gsm  ulaw  alaw  g726 adpcm  slin lpc10  ilbc g726aal2  g722 slin16 testlaw slin12 slin24 slin32 slin44 slin48 slin96 slin192
      gsm     - 15000 15000 15000 15000  9000 15000 15000    15000 17250  17000   15000  17000  17000  17000  17000  17000  17000   17000
     ulaw 15000     -  9150 15000 15000  9000 15000 15000    15000 17250  17000   15000  17000  17000  17000  17000  17000  17000   17000
     alaw 15000  9150     - 15000 15000  9000 15000 15000    15000 17250  17000   15000  17000  17000  17000  17000  17000  17000   17000
     g726 15000 15000 15000     - 15000  9000 15000 15000    15000 17250  17000   15000  17000  17000  17000  17000  17000  17000   17000
    adpcm 15000 15000 15000 15000     -  9000 15000 15000    15000 17250  17000   15000  17000  17000  17000  17000  17000  17000   17000
     slin  6000  6000  6000  6000  6000     -  6000  6000     6000  8250   8000    6000   8000   8000   8000   8000   8000   8000    8000
    lpc10 15000 15000 15000 15000 15000  9000     - 15000    15000 17250  17000   15000  17000  17000  17000  17000  17000  17000   17000
     ilbc 15000 15000 15000 15000 15000  9000 15000     -    15000 17250  17000   15000  17000  17000  17000  17000  17000  17000   17000
 g726aal2 15000 15000 15000 15000 15000  9000 15000 15000        - 17250  17000   15000  17000  17000  17000  17000  17000  17000   17000
     g722 15600 15600 15600 15600 15600  9600 15600 15600    15600     -   9000   15600  17500  17000  17000  17000  17000  17000   17000
   slin16 14500 14500 14500 14500 14500  8500 14500 14500    14500  6000      -   14500   8500   8000   8000   8000   8000   8000    8000
  testlaw 15000 15000 15000 15000 15000  9000 15000 15000    15000 17250  17000       -  17000  17000  17000  17000  17000  17000   17000
   slin12 14500 14500 14500 14500 14500  8500 14500 14500    14500 14000   8000   14500      -   8000   8000   8000   8000   8000    8000
   slin24 14500 14500 14500 14500 14500  8500 14500 14500    14500 14500   8500   14500   8500      -   8000   8000   8000   8000    8000
   slin32 14500 14500 14500 14500 14500  8500 14500 14500    14500 14500   8500   14500   8500   8500      -   8000   8000   8000    8000
   slin44 14500 14500 14500 14500 14500  8500 14500 14500    14500 14500   8500   14500   8500   8500   8500      -   8000   8000    8000
   slin48 14500 14500 14500 14500 14500  8500 14500 14500    14500 14500   8500   14500   8500   8500   8500   8500      -   8000    8000
   slin96 14500 14500 14500 14500 14500  8500 14500 14500    14500 14500   8500   14500   8500   8500   8500   8500   8500      -    8000
  slin192 14500 14500 14500 14500 14500  8500 14500 14500    14500 14500   8500   14500   8500   8500   8500   8500   8500   8500       -

Attaching sip.conf…its only for outbound

sip conf.txt (3.0 KB)

Attaching output for CLI

sip CLI debug.txt (49.2 KB)
sip debug

7

Device ‘800’ has negotiated gsm and ulaw. You only have g729 allowed on voxbeam which is negotiated. Asterisk has to transcode and can’t, as g729 is not an included codec for translation.

1 Like
8

As @jcolp said device has negotiated only (gsm|ulaw)

Capabilities: us - (gsm|ulaw), peer - audio=(gsm|ulaw|alaw|speex|ilbc)/video=(nothing)/text=(nothing), combined - (gsm|ulaw)

while on the voxbeam configuration you have this
[voxbeam]

host=eu.voxbeam.com
disallow=all
allow=g729

I suggest you replace g729 by ulaw

1 Like
9

is there any way I can g729 in codec translation? Do I have to install it seperately?

10
11

I believe there are a free G729 codecs which are are only supported in Asterisk for the multiple channels as 'Pass through mode, will that work if I install on my server, as I have CISCO IP Phones which supports g729?

12

Not sure but I think promote free g729 codec might violate some of the rule of this forum

13

Sorry, I didn’t ment to promote…was just trying to resolve my issue

BTW those codecs I mentioned are just for “pass through mode”, not oragination or termination of call.

14

The only so called “free” G.729 codec, that I have heard of, has a void licence, as it is based on sample code which has a no commercial use restriction, but purports to be under the GPL which has a no-no commercial use restriction restriction, and a clause that makes the whole licence void if you can’t fully comply.

Any free implementation could only be legal in countries that do not honour US software patents.

1 Like
15

No I can’t use G.729 then as we have to honor US software patents. I guess I should go for something free like G.711? Please advice

16

G.711 is what the PSTN itself uses, within a continent. Note that there are two variants of G.711. You want mu-law for the North America and Japan and A-law for the rest of the world. There will be a slight degradation, when it gets transcoded for the PSTN, if you use the wrong one, although I notice that most Cisco installations in the UK use the wrong one.