Call from External to AA

Unanswered Question
Dec 31st, 2009
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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

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Marwan ALshawi Thu, 12/31/2009 - 05:28
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try to add the follwoing dial peer and see if it helps or not


dial-peer voice 90 voip
incoming called-number .

codec g711ulaw


good luck

if helpful Rate

ricardorojas123 Thu, 12/31/2009 - 05:47
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I change the settings to dial-peer called dial-peer voice 1902 voip (dial peer to the AA), codec and incoming number, but dont work .

ricardorojas123 Mon, 01/04/2010 - 12:01
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I noticed that the calls between the internal IP Cisco Phone to the Auto Attendant in the Cisco Unity Express are established using the g711ulaw codec, and the codec used between the external call to internal IP Cisco Phone is g711alaw . Is possible change the codec g711ulaw to g711alaw in he connection from the CUCM to CUE ?


Example, actual codec used:


ITSP  --> g711alaw  --> (CUCME) ---> g711ulaw ---> Auto Attendant (CUE)

Cisco IP Phone registered in the CUCME ---> g711ulaw ----> Auto Attendant (CUE)

Marwan ALshawi Mon, 01/04/2010 - 14:55
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you can by using incoming voip dialpeer same concept of the above one


lets say from cucm ip phnes dial cue with number 1000


in the voice gateway where cue reside:


dial-peer voice 99 voip

incoming called-number 100

codec g711ulaw


and you already have outgoing dial-peer with the command destination pattern to send the the call to the right destination


good luck

jon.aril.antonsen Wed, 10/13/2010 - 03:21
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Hello ricardorajas123


Did you manage to solve this problem?


I have exactly the same problem as you had, and I'm pulling my hair on this one...


I hope you have an answer.


regards,

Jon

ADAM CRISP Wed, 10/13/2010 - 08:03
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Hi


As you need to get an incoming SIP call routed as a SIP call to the CUE, also check to see if you have allowed this

i.e

voice sevice voip

allow connections sip to sip


I would also recommend configuring the CUE dial-peer as a b2bua.


If you are switching on the sip-sip feature, please make sure you don't inadventantly make your router into an open SIP gateway.


Adam

jon.aril.antonsen Wed, 10/13/2010 - 08:07
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Hi Adam.


Cheers for the input.

All the settings you mentioned are set, but no luck :-(



Jon

jon.aril.antonsen Wed, 10/13/2010 - 08:01
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Hi Dilip.


I configured the transcode a bit earlier today, and now the call is forwarded to VM/AA. But the problem now is that I do NOT hear the VM/AA prompt...just silence. With debug I can see that the call is active.



Jon

dksingh Wed, 10/13/2010 - 08:12
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Hmm..I think u mean, you do *NOT* hear CUE prompts. The first thing you want to check is the RTP legs and if

xcoding is getting invoked. Following will help:


sh voip rtp connect

sh sccp connection


FYI, for ulaw<->alaw conversion you'd need universal xcoders:


http://www.cisco.com/en/US/docs/ios/12_4t/12_4t15/it_unitr.html


Another thing to note is for xcoding to be invoked, you need to hardcode codecs on both inbound and outbound

dialpeers rather than using a voice class-codec command.

Are u comfortable with capturing debugs and sharing your config in case we need to troubleshoot it further?


DK



jon.aril.antonsen wrote:


Hi Dilip.


I configured the transcode a bit earlier today, and now the call is forwarded to VM/AA. But the problem now is that I do hear the VM/AA prompt...just silence. With debug I can see that the call is active.



Jon

jon.aril.antonsen Wed, 10/13/2010 - 08:18
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Hi.


I will look into this right away.


It will be no problem sharing debugs and configs. This is just in a lab enviroment...so no secrets here :-)


When you say both in an put dial peers I rekin you refer to the VM/AA dial-peers?


I will post the result shortly :-)


Jon.

jon.aril.antonsen Wed, 10/13/2010 - 08:29
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Hi again.


This is what i got from the show commands. The commands issued while having active call to the AA PSTN number.


UC520#sh voip rtp conn
VoIP RTP active connections :
No. CallId     dstCallId  LocalRTP RmtRTP LocalIP            RemoteIP                              
1   1337       1338       19512    22430  MY_OUTSIDE_IP     SIP_GW_IP                         
2   1338       1337       16384    20844  10.1.10.2          10.1.10.1                             
3   1339       1340       19304    2000   10.1.10.2          10.1.1.1                              
4   1341       1340       17082    2000   10.1.10.2          10.1.1.1                              
Found 4 active RTP connections


UC520#sh sccp conn
sess_id    conn_id      stype mode     codec   sport rport ripaddr

7          7            xcode sendrecv g711u   17722 2000  10.1.1.1
7          8            xcode sendrecv g711a   17490 2000  10.1.1.1

Total number of active session(s) 1, and connection(s) 2


I'm also quite sure the codecs are configured correctly. But still just silence.


Jon.

dksingh Wed, 10/13/2010 - 08:39
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Yep, xcoder is getting invoked it seems but those IP addr.

don't ring a bell in the absence of config.

If u can, please go ahead and capture following debugs for

one call from SIP trunk/PSTN to CUE/AA:



deb ccsip mess

deb ccsip err

deb voip ccapi inout


Note: Please capture debugs in a buffer ; configure following

         before enabling/capturing debugs:


         conf t

         service time deb date msec

         service sequence

          no logg con

          logg mon warn

          logg buffer 5000000 debug

          no logging rate


Issue clear log just before making the test call and once done

issue term len 0 and capture sh log output in a text file.



Please include calling/called num, call flow and config (sh run)

Pl. capture sh voip rtp connect and sh sccp connect for the same

call as well.


DK

jon.aril.antonsen Wed, 10/13/2010 - 09:02
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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 Wed, 10/13/2010 - 09:23
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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 Wed, 10/13/2010 - 09:37
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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 Wed, 10/13/2010 - 09:41
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nice!

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

jon.aril.antonsen Wed, 10/13/2010 - 09:54
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I don't think I can mark it as answered since it was not me that started this post.


Jon

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