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No RTP stream from ISP SIP via H.323 Gateway to SCCP Phone (CUCM 8.0)

I've got next schema:

PSTN - ISP - (SIP trunk) - AS5350XM - (H.323) - CUCM8.0 - (SCCP) - IP Phone

AS

There is a problem with incoming call from PSTN.

This is how the progress of the call looks like:

Call comes to GW

GW sends it to CUCM. CUCM translate number to internal.

IP Phone rings.

I hook up the handset and hear nothing. Packet capture shows that some udp traffic goes between the phone and the GW. 

134 1.174445000 10.20.8.46 10.20.8.10 UDP 214 Source port: 31926  Destination port: 17774

At the same time on the calling side I continue to hear ring-back.

After few seconds call drops.

 

H.245 Event Debug after hooking up shows next:

hq_as5350xm_gw1#changing from CONNECTING state to CONNECTED state

Sep 25 17:08:16.145: h323chan_chn_process_read_socket: fd=3 of type CONNECTED has data

Sep 25 17:08:16.145: h323chan_recvdata: No Data on fd=3
PROCESS_READ: FAILED/NOT COMPLETE,rc 10, fd=3

 

After some timeout and droping:

Sep 25 17:11:50.478: h323chan_chn_process_read_socket: fd=2 of type CONNECTED has data

Sep 25 17:11:50.482: h323chan_close: TCP connection from fd=2 closed
Sep 25 17:11:50.482: h323chan_close: TCP connection from fd=3 closed

 

And ccsip messages debug respectively:

Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 193.***.***.145:5060;branch=z9hG4bKwqpxz8y1cxyvyutt7v7pmq7vx
From: <sip:4436****1@62.***.***.26;user=phone>;tag=sbc0407q77put8w-CC-22
To: <sip:322*****4@62.***.***.266;user=phone>;tag=8C2BA830-20B0
Date: Thu, 25 Sep 2014 17:11:38 GMT
Call-ID: isbcv1pz8by8ym8pwyuqxpmbqhmqhxph88px@SoftX3000
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=47
Content-Length: 0


Sep 25 17:11:50.490: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:

hq_as5350xm_gw1#ACK sip:322*****4@62.***.***.26:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 193.***.***.145:5060;branch=z9hG4bKwqpxz8y1cxyvyutt7v7pmq7vx
Call-ID: isbcv1pz8by8ym8pwyuqxpmbqhmqhxph88px@SoftX3000
From: <sip:4436****1@62.***.***.26;user=phone>;tag=sbc0407q77put8w-CC-22
To: <sip:322*****4@62.***.***.26;user=phone>;tag=8C2BA830-20B0
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

 

 

Incoming dial-peer:

dial-peer voice 43 voip
 translation-profile outgoing LV_KS-SIP_out
 session protocol sipv2
 session target ipv4:193.***.***.145
 incoming called-number 322*****4
 voice-class codec 1
 voice-class sip bind control source-interface GigabitEthernet0/1.202
 voice-class sip bind media source-interface GigabitEthernet0/1.202

 

 

Outgoing Dial-peer

dial-peer voice 103 voip
 destination-pattern 322*****4
 session target ipv4:10.20.8.6
 voice-class codec 1

 

 

Interface to ISP

interface GigabitEthernet0/1.202
 encapsulation dot1Q 202
 ip address 62.***.***.26 255.255.255.248

Interface to CUCM

interface GigabitEthernet0/0
 ip address 10.20.8.10 255.255.255.0
 duplex auto
 speed auto
 negotiation auto
 h323-gateway voip interface
 h323-gateway voip bind srcaddr 10.20.8.10

 

 

voice service voip
 ip address trusted list
  ipv4 10.0.0.3
  ipv4 10.0.0.4
  ipv4 193.***.***.145
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol pass-through g711ulaw
 sip
!

 

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

 

 

As I understand I missed something in theory about SIP and H.323 interoperability. Please, help me to find my gap.

 

 

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