Cannot determine best translation path: one capability supports no formats

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:


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

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

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.

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.

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