Cannot determine best translation path: one capability supports no formats

Asterisk Asterisk Support
1

I’ve been circling the drain on this one for a minute and need some help understanding exactly what’s going on.
I have an Asterisk instance (18.17.1) running chan_sip which sends/receives SIP Messages of a certain Content-Type.
Receiving and sending SIP messages works just fine; since they come in out of context and without a specific extension so I have to catch them with an s rule (logs sanitized):

exten => s,1,Goto(<context>,<extension>,1)

However, another part of that process is originating an outbound SIP channel, which is where I’m running into this error:

…
[May 31 14:23:45] VERBOSE[301917][C-00000001] pbx.c: Executing [<extension>@<context>:18] Originate(“Message/ast_msg_queue”, “SIP/+<outbound_number>@<outbound_peer>!!+<caller_id>@<sip_host>,exten,<context>,<extension>,19”) in new stack
[May 31 14:23:45] VERBOSE[301917][C-00000001] netsock2.c: Using SIP RTP CoS mark 5
[May 31 14:23:45] VERBOSE[302058] dial.c: Called +<outbound_number>@<outbound_peer>!!+<caller_id>@<sip_host>
[May 31 14:23:48] VERBOSE[301934][C-00000001] res_rtp_asterisk.c: 0x7fca3c011f40 – Strict RTP learning after remote address set to: <outbound_peer>:26834
[May 31 14:23:48] VERBOSE[302058] dial.c: SIP/<outbound_peer>-00000000 is making progress
[May 31 14:23:48] VERBOSE[302058] res_rtp_asterisk.c: 0x7fca3c011f40 – Strict RTP switching to RTP target address <outbound_peer>:26834 as source
[May 31 14:23:49] VERBOSE[302058] dial.c: SIP/<outbound_peer>-00000000 is ringing
[May 31 14:23:49] VERBOSE[302058] dial.c: SIP/<outbound_peer>-00000000 is ringing
[May 31 14:23:53] VERBOSE[302058] res_rtp_asterisk.c: 0x7fca3c011f40 – Strict RTP learning complete - Locking on source address <outbound_peer>:26834
[May 31 14:24:16] NOTICE[301917][C-00000001] Ext. <extension>: “RINGING”
[May 31 14:24:17] ERROR[302132] translate.c: Cannot determine best translation path since one capability supports no formats
[May 31 14:24:17] WARNING[302132] channel.c: No path to translate from SIP/<outbound_peer>-00000001 to Message/ast_msg_queue

This is the only logging I’m getting about it; nothing actually codec-specific. sip.conf is configured to allow both ulaw and alaw:

…
disallow=all
allow=ulaw
allow=alaw
…

But I don’t think it’s a codec mismatch issue because a) I’ve been interacting with this outbound peer for over a year using those codecs with no problem and b) there’s nothing codec-specific actually being logged.
I’m wondering if this is a problem with me trying to originate a call from Message/ast_msg_queue? Am I not understanding what I can ‘do’ at this point? The end goal is to put that originated channel into a stasis app.

Thanks in advance

2

It’s probably trying to use the codecs on the channel invoking the Originate dialplan application. You can try using the ‘C’ option to explicitly specify instead[1].

[1] Originate - Asterisk Documentation

1 Like
3

That was the problem, thank you. The codec configuration was only located in a peer template definition, so there was no way to match originate’s default of slin with the allowed ulaw/alaw.

4

They shouldn’t come in out of context, because:

  1. you should have specified a context:
  1. All dialplan execution has a context, which will be the default one, if nothing else is specified.
5

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