Why is ulaw the native format

I’m on a bit of a mission to reduce the amount of warning and unnecessary messages in our log files. We pay close attention to them but they fill up with massive amount of unnecessary stuff, and im not willing to set the log level higher - some of the warnings are actually important. But we write about 1 million lines per day, so i want to try reduce this.

Right now the main culprit is:

I don’t understand the need to warn about this - it seems pointless. Also why would the channel be on ulaw. In no place do we use ulaw. Isn’t that the Canadian standard anyway? If anything i would have thought it would say alaw?

This is my sip conf as loaded:

[code]sip show settings

Global Settings:

UDP Bindaddress:
TCP SIP Bindaddress:
TLS SIP Bindaddress: Disabled
Videosupport: Yes
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: Yes
SIP domain support: No
Realm. auth: No
Our auth realm somevoipco.com
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: Some Voip PBX
SDP Session Name: Asterisk PBX 11.19.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: unknown
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: Yes
T.38 EC mode: Redundancy
T.38 MaxDtgrm: 400
SIP realtime: Enabled
Qualify Freq : 60000 ms
Q.850 Reason header: No

Network QoS Settings:

IP ToS RTP audio: EF
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 3
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: Yes
Jitterbuffer forced: No
Jitterbuffer max size: 200
Jitterbuffer resync: 1000
Jitterbuffer impl: fixed
Jitterbuffer log: No

Network Settings:

SIP address remapping: Disabled, no localnet list
Externaddr: (null)
Externrefresh: 10

Global Signalling Settings:

Codecs: (alaw|g729)
Codec Order: g729:20,alaw:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 60
RTP Timeout: 120
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70

Default Settings:

Allowed transports: UDP,TCP
Outbound transport: UDP
Context: from-outside
Record on feature: automon
Record off feature: automon
Force rport: Auto (No)
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: Never
Tone zone:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk

Realtime SIP Settings:

Realtime Peers: Yes
Realtime Regs: No
Cache Friends: Yes
Update: Yes
Ignore Reg. Expire: No
Save sys. name: Yes
Auto Clear: 60 (Enabled)


Mu-law isn’t the native format, in this example.

It’s a warning because speech quality has been compromised because a 20ms gap has been left in the data.

I believe the current codecs are reported in “core show channel…”, not “sip show channel…”, as this message comes from the Asterisk core.

One would need “sip set debug on” and possibly “core set debug 5 chan_sip” to stand any chance of understanding why you had an incompatible frame. One would also need to know if you are running in G.729 in pass through, or fully licensed mode.