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g729r8 Codec Offer g711ulaw Internal Server Error

MrButton
Beginner
Beginner

There is this one SIP Service Provider that I send SIP invites from a 4431 ISR to another 4431 then to SIP SP fails.  I get an internal server error from them.  As soon as the first Dial-Peer fails it works with another SIP Service Provider using g711ulaw b/c that is what is negotiated.  I am using the g711ulaw as the primary preference in the config from the originating router, but this SIP invite starts off with g729 when it reaches the edge cube connected to the SIP trunks hand offs.

This will continue to ring even after answering, then just fail.  As soon as the called # hangs up (my cell phone) the edge 4431 fails over to another Dial-Peer that has ^9.T destination pattern.

I do a sh call act voice compact:

g729r8 pre- VOIP

In the originating voice gateway:

voice service voip
ip address trusted list
ipv4 SUBNETWORK/8
qsig decode
address-hiding
mode border-element license capacity 5
media disable-detailed-stats
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
modem passthrough nse codec g711ulaw
h323
sip
bind control source-interface GigabitEthernet0/0/0
bind media source-interface GigabitEthernet0/0/0
header-passing
error-passthru
registrar server expires max 3600 min 3600
early-offer forced <--------------------------------------------
midcall-signaling passthru
no call service stop

voice class codec 1001
codec preference 1 g711ulaw <--------------------------------------------

dial-peer voice 346 voip
description outgoing to JAO-Cube
preference 3
session target ipv4:DEST_IP_OF_CUBE_TO_SP
destination e164-pattern-map 1022
codec g711ulaw <------------------------------------------------------
fax rate 14400 bytes 48
ip qos dscp cs3 signaling
no vad
!

Any help would be appreciated.

I have a TAC case open but TAC seems to be really busy.  I rarely can get a chance with the person assigned to the case to do testing with me.  For now Ive been just trying on my own.  The originating voice gateway is setup with an FXO card POTS.  I want to route a Dial-Peer when POTS is busy to another CUBE using the primary SIP service provider for external network calls. It works with the fail over SIP.  Two different service providers. 

1 Accepted Solution

Accepted Solutions

MrButton
Beginner
Beginner

resolved.  SDP header SIP profile in CUCM. applied rel1xx and early offer. good now.

View solution in original post

2 Replies 2

b.winter
VIP Advocate VIP Advocate
VIP Advocate

Hi,

please form clear and structured sentences. The only thing I'm guessing from all the text you wrote is, that you probably have a codec issue, but the text is very confusing.
Which entities are involved?
What is the telephony system?
What is the call flow?
Describe the setup (Who is the first ISR, who is the second, where does the call come from, ...)
Provide the full config (without any sensitive data like pwds, username, ...)(nobody can work with junks of config)
Provide the output with timestamps and numbers involved from a complete call:
debug ccsip messages
debug voice ccapi ind 1
debug voice ccapi ind 2
debug voice ccapi ind 74

In your text, you write something about SIP provider, but the dial-peer you are showing is not a SIP dial-peer?
Also, you are configuring the voice class codec 1001, but then in the dial-peer you set the codec static.
If you don't use SRST / CME on the router, then you don't need the command "registrar server expires max 3600 min 3600"

MrButton
Beginner
Beginner

resolved.  SDP header SIP profile in CUCM. applied rel1xx and early offer. good now.

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