[Asterisk-Users] ULaw not negotiating

Paul Rodan asterisk at glitch.cc
Fri Jan 14 15:50:07 MST 2005


Ok,

 

My provider is sending a call to me via ULaw but Asterisk isn't picking up
on this, I've only allowed ulaw, I disallow=all and then allow=ulaw in my
sip.conf and that's the only thing I allow, but when my provider sends me
the requests, I get an error about No Compatible Codecs:

 

 

17 headers, 8 lines

Using latest request as basis request

Sending to 67.19.245.213 : 5060 (non-NAT)

Found RTP audio format 18

Found RTP audio format 101

Peer audio RTP is at port 38.114.20.207:28442

Found description format G729

Found description format telephone-event

Capabilities: us - 0x4(ULAW), peer - audio=0x100(G729A)/video=0x0(EMPTY),
combined - 0x0(EMPTY)

Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
0x1(G723)

Jan 14 17:29:55 WARNING[81922]: chan_sip.c:2820 process_sdp: No compatible
codecs!

 

What throws me off is the description format of G729. They said they used to
be sending in ULaw and G729, but then I had him turn off G729 all together.
But this sip debug doesn't confirm that, I see g729 mentioned several times.
But I do see ULaw mentioned in there as well as well as 0x4, under
Capabilities. So why isn't Asterisk accepting it?

 

Any help with this debug would be immensely helpful!

 

 

I also have BroadVoice which sends me calls ULaw and works fine, so I called
that number and captured the inbound with sip debug and saw this:

 

14 headers, 9 lines

Using latest request as basis request

Sending to 147.135.4.128 : 5060 (non-NAT)

Found RTP audio format 0

Found RTP audio format 101

Peer audio RTP is at port 147.135.4.128:14664

Found description format PCMU

Found description format telephone-event

Capabilities: us - 0x4(ULAW), peer - audio=0x4(ULAW)/video=0x0(EMPTY),
combined - 0x4(ULAW)

Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
0x1(G723)

Found peer 'broadvoice'

Looking for 5555551212 in incoming

list_route: hop:
<sip:5555551313 at 147.135.4.128:5060;bvoice=ACME-asdfadf2dfsa3;ep=147.135.4.12
9;transport=udp>

Transmitting (no NAT):

SIP/2.0 100 Trying

Via: SIP/2.0/UDP
147.135.4.128:5060;branch=jkjk245kjelkjelkj2435sadflkj435.1sr

From: "TEST
PHONE"<sip:5555551313 at 147.135.4.129;user=phone;bvoice=ACME-06t5tpji5ub7e>;ta
g=SD50vt601-662260634-1105742161664

To: "Deon
Rodden"<sip:5555551212 at sip.broadvoice.com;user=phone>;tag=as59299ec2

Call-ID: SD50vt601-8b297b5d0b4543648439f18f9eba5903-js19002

CSeq: 967783297 INVITE

User-Agent: CSCO/7

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

Contact: <sip:5555551212 at 123.123.123.123>

Content-Length: 0

Answering with preferred capability 0x4(ULAW)

Reliably Transmitting (no NAT):

 

 

Best Regards,

Paul

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