Call from External to AA

[ricardorojas123]

I have a trunk SIP connected to CCME, this trunk SIP is used to process incoming and outgoing calls. The incoming and outgoing calls works fine. The problem is the AutoAttendant and VoiceMail. Internally the extensions can hear the Auto Attendant and when a extension does not answer, the prompt of the voice mail is heard, but from the external call to AutoAttendant, the prompt is not heard, the same for the voice mail, the prompts are not heard.

Topology:

(Router_2800) --->SIP trunk ---->  (ISP)

This is the configuration:

voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8

dial-peer voice 1900 voip
description *InternalExtensions*
destination-pattern 555....
voice-class codec 1
session protocol sipv2
session target ipv4:172.17.30.1
dtmf-relay sip-notify

dial-peer voice 1901 voip
description ***CUE_Voicemail***
destination-pattern 1901
session protocol sipv2
session target ipv4:172.17.30.2
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 1902 voip
description *AutoAttendant*
destination-pattern 1902
voice-class codec 1
session protocol sipv2
session target ipv4:172.17.30.2
dtmf-relay sip-notify
no vad

dial-peer voice 2000 voip
tone ringback alert-no-PI
description *External Calls*
destination-pattern 9143T
voice-class codec 1
session protocol sipv2
session target ipv4:172.19.25.2
dtmf-relay rtp-nte

[jon.aril.antonsen]

Hi.

The capture is done. The files are attached.

Called number is 52223120. AA number is 398

I hope this is what you need.

Jon

EDIT: Attached files deleted.

Message was edited by: jon.aril.antonsen

[dksingh]

hmm...something weird...we are sending back g711ulaw in the 200 OK to the SIP provider:

002270: Oct 13 15:57:19.309: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
----------------snip----------------------------
v=0
o=CiscoSystemsSIP-GW-UserAgent 7159 4228 IN IP4 79.160.246.126
s=SIP Call
t=0 0
m=audio 17508 RTP/AVP 0 101  <-------------- (0=g711ulaw)
c=IN IP4 79.160.246.126

whereas in the initial invite we receive:

m=audio 23154 RTP/AVP 8 101    (8=g711alaw)

Can u hardcode codec in the inbound dialpeer and test ?

dial-peer voice 1006 voip
description ** AA from SIP Trunk (Auto Attendant 1)**
  voice-class codec 1  <--remove

   codec g711alaw  <--add

Also, have u tried getting rid of ACL from loop0 and vlan100 for testing?

I also notice that u r not using universal xcoders

dspfarm profile XX transcode universal  <------pl. see previous doc I mentioned.

[jon.aril.antonsen]

Hi :-)

And just like magic...it works :-) It was the hardcode for the codec.

Thank you so much, I really appreciate your help on this one.

So now I will keep on learning this box.....and probably post my next problem soon :-p

Jon

[dksingh]

nice!

Please rate the post and mark it answered when u get a moment.

[jon.aril.antonsen]

I don't think I can mark it as answered since it was not me that started this post.

Jon