[asterisk-users] Does Early Media have to be Ulaw?

[Douglas Garstang]

I have this in sip.conf:

 

[general]

context=default

allowoverlap=no

bindport=5060

bindaddr=0.0.0.0

srvlookup=yes

progressinband=yes

 

[19256002182]

type=friend

username=19256002182

callerid="Test hone 1" <+19256002182>

host=dynamic

canreinvite=no

secret=password

context=test

disallow=all

allow=g729

 

[level3]

type=peer

host=xxx.yyy.16.99

context=default

insecure=port

dtmfmode=rfc2833

canreinvite=yes

qualify=yes

disallow=all

;allow=ulaw

allow=g729

 

Level 3 sends early media...

 

<--- Transmitting (no NAT) to xxx.yyy.34.210:5061 --->

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP
xxx.yyy.34.210:5061;branch=z9hG4bK-tenor-d802-22d2-004d;received=xxx.yyy
.34.210

From: <sip:19256002182 at xxx.yyy.34.195>;tag=d80222d2-27

To: <sip:13033372500 at xxx.yyy.34.195>;tag=as4fe079a5

Call-ID: call-71EEDB19-F102-2A10-0B0C-1B at xxx.yyy.34.210

CSeq: 2 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: <sip:13033372500 at xxx.yyy.34.195>

ontent-Type: application/sdp

Content-Length: 261

 

v=0

o=root 2235 2235 IN IP4 xxx.yyy.34.195

s=session

c=IN IP4 xxx.yyy.34.195

t=0 0

m=audio 10484 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

 

and Asterisk responds on the console with:

 

[Jun 22 10:06:03] WARNING[32573]: channel.c:2882 set_format: Unable to
find a codec translation path from g729 to slin

[Jun 22 10:06:03] WARNING[32573]: indications.c:121 playtones_alloc:
Unable to set 'SIP/19256002182-096ac918' to signed linear format (write)

 

This doesn't happen when progressinband=no. It almost seems like
Asterisk has to do early media as G711 only. Is that the case???

 

Doug.

 

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