Outbound Call Error es_rtp_asterisk.c: Unknown RTP codec 95 received from '192.168. from local pc

I am getting this error when trying to make outgoing call on elastix free pabx using zoiper soft phone can any 1 pls help me to sort this out…mush appriciated

[2019-04-09 01:00:23] VERBOSE[4034][C-00000017] pbx.c: – Executing [00770259128@from-internal:1] ResetCDR(“SIP/1004-00000019”, “”) in new stack

[2019-04-09 01:00:23] VERBOSE[4034][C-00000017] pbx.c: – Executing [00770259128@from-internal:2] NoCDR(“SIP/1004-00000019”, “”) in new stack

[2019-04-09 01:00:23] VERBOSE[4034][C-00000017] pbx.c: – Executing [00770259128@from-internal:3] Progress(“SIP/1004-00000019”, “”) in new stack

[2019-04-09 01:00:23] VERBOSE[4034][C-00000017] pbx.c: – Executing [00770259128@from-internal:4] Wait(“SIP/1004-00000019”, “1”) in new stack

[2019-04-09 01:00:23] NOTICE[4034][C-00000017] res_rtp_asterisk.c: Unknown RTP codec 95 received from ‘192.168.X.X Local pc IP:8000’

[2019-04-09 01:00:24] VERBOSE[4034][C-00000017] pbx.c: – Executing [00770259128@from-internal:5] Progress(“SIP/1004-00000019”, “”) in new stack

[2019-04-09 01:00:24] VERBOSE[4034][C-00000017] pbx.c: – Executing [00770259128@from-internal:6] Playback(“SIP/1004-00000019”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack

[2019-04-09 01:00:24] VERBOSE[4034][C-00000017] file.c: – <SIP/1004-00000019> Playing ‘silence/1.gsm’ (language ‘en’)

[2019-04-09 01:00:25] VERBOSE[4034][C-00000017] file.c: – <SIP/1004-00000019> Playing ‘cannot-complete-as-dialed.gsm’ (language ‘en’)

[2019-04-09 01:00:28] VERBOSE[4034][C-00000017] file.c: – <SIP/1004-00000019> Playing ‘check-number-dial-again.gsm’ (language ‘en’)

[2019-04-09 01:00:29] VERBOSE[4034][C-00000017] pbx.c: == Spawn extension (from-internal, 00770259128, 6) exited non-zero on ‘SIP/1004-00000019’

[2019-04-09 01:00:29] VERBOSE[4034][C-00000017] pbx.c: – Executing [h@from-internal:1] Hangup(“SIP/1004-00000019”, “”) in new stack

[2019-04-09 01:00:29] VERBOSE[4034][C-00000017] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/1004-00000019’

Looks like it’s working to me.

Soft phone placed a call to the PBX which answered, played some media and hung up.

The only error I see is

Which sounds legit to me, 95 isn’t a defined codec so your softphone is doing something non-standard.

what is the option to sort this out…I get error message when i try to ping the SIP Server IP which is ISP’s 172.XX.X.X, I put trace Route

root@localhost ~]# tracert 172.1X.X.X

traceroute to 172.1X.X.X (172.1X.X.X), 30 hops max, 40 byte packets

1 (192.168.X.XX)“PABX Server” 3000.797 ms !H 3000.800 ms !H 3000.800 ms !H

need an assistant pls

Fix the peer. Assuming you are using a supported version, Asterisk is not at fault and there are no options to work round this sort of error with a peer.

I think you are actually quoting an RTP error message, in which case peer tried to use a codec that wasn’t “agreed” in the SDP exchange, butj it would also be invalid if it was specified n the SDP exchange

Your traceroute problem is quite likely the use of firewall rules that frustrate the use of ICMP, although it could be a routing configuration problem or firewall rules that frustrate the UDP probe used for traceroute.

The RTP codec 95 error isn’t an error, it’s a notice and is not fatal. It’s not the source of the problem.

The problem seems more to be in the dialplan produced by Elastix PBX or its configuration.

It was a poor word choice in my reply, you are correct that the RTP codec message was only a warning.

Hi…thanks for ure reply if my pilot number is 0112332247 how to the dial patten should come isp said the dial patten without 0 rest number

The easiest way is to log the protocol, or even look at the error message, to see what your ITSP is actually sending as the user part of the request URI.

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