09-21-2010 12:19 PM - edited 03-16-2019 12:54 AM
I have managed to register a third party SIP client on a Cisco Call Manager Express v7.0 successfully. Having done that, I am able to make calls from the SIP client to the Cisco phones that are registered in the CME and vice-versa as well as calls to the "outside world" through the Call Manager Express Configuration.
However, the following strange thing occurs:
When I am trying to make calls from the SIP client to a Cisco phone that is registered to a Cisco Unified Call Manager (6.1.3 server), the Cisco phone rings, I answer the call on the phone, the phone seems to be connected (counting seconds) but on the SIP client side it keeps waiting for answer .
A couple of seconds later, it seems that the SIP client tries to call again (the Cisco phone accepts a second call) but again with no success until the call is dropped.
The other way around (Cisco of the CUCM to SIP client of CME) works just fine.
Obviously Cisco to Cisco calls between the two Call Managers work normally.
Please see below part of the CME configuration:
voice service voip
no notify redirect ip2ip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
h323
h225 h245-address on-connect
no call service stop
call start slow
sip
session transport tcp
registrar server
midcall-signaling passthru
Do you think that this may be any kind of SIP authentication or configuration that may be missing?
Thanks in advance for any suggestions
09-22-2010 07:06 AM
It's possible that it may be a codec mismatch. Can you check which dial-peers are matched and what codecs are specified?
-Felipe
09-22-2010 07:32 AM
Hello Felipe,
Thank you for the reply,
From the CME side I have the following (we use a primary and a secondary CUCM)
dial-peer voice 100 voip
description *** CUCM (Primary) ***
destination-pattern 6[1-9]..
session target ipv4:CUCM_IP_address
dtmf-relay h245-alphanumeric
codec g711ulaw
ip qos dscp cs5 media
no vad
!
dial-peer voice 101 voip
description *** CUCM (Secondary) ***
preference 1
destination-pattern 6[1-9]..
session target ipv4:CUCM_IP_address
dtmf-relay h245-alphanumeric
codec g711ulaw
ip qos dscp cs5 media
no vad
!
On the other hand, on the Cisco Unified Call Manager 6.1.3 , while checking on the Region, the codec that is used for the communication between the 2 sites is denoted as G.711. I am not sure if I need to check something else.
What is really confusing for me is that it works only from the Cisco to the SIP and not the other way around, and most important that when calling from SIP (CME) the Cisco phone (CUCM) rings (that is the call is routed) and shows connected.
Thanks again for your assistance on this
George
09-22-2010 07:34 AM
Can you collect the following debug output.
debug voip ccapi inout
debug ccsip message
Please note the calling and called party numbers.
Also, include the full configuration of the gateway.
-Felipe
09-22-2010 08:25 AM
Hello Felipe,
Attached you may find the sh run config of the CME along with the debug file.
Calling Details:
The SIP client with extension 5730 calls the Cisco phone with extension 6638.
CME IP address 10.20.10.10
SIP client: 5730 (IP address 10.1.1.25)
CUCM IP address 10.2.1.250
Cisco phone 6638 (10.2.1.38)
Best regards,
09-22-2010 12:33 PM
I'll need you to collect another set of debugs.
debug h225 asn1
debug h245 asn1
debug voip ccapi inout
debug ccsip message
The call is disconnected by CUCM so, if possible, a set of detailed level traces would also help.
-Felipe
09-23-2010 01:33 AM
Hello Felipe,
Please find attached the debug files that you suggested.I hope these work for troubleshooting.
Again I use the same calling scenario:
The SIP client with extension 5730 calls the Cisco phone with extension 6638.
CME IP address 10.20.10.10
SIP client: 5730 (IP address 10.1.1.25)
CUCM IP address 10.2.1.250
Cisco phone 6638 (10.2.1.38)
Please let me know if you think that something is missing,
Thanks once more,
George
09-23-2010 06:08 AM
Please upload the CUCM traces. The files collected from CUCM are not the traces but the log files for when the traces were collected. They should either be zipped or in a folder named after the CUCM nodes.
-Felipe
09-23-2010 06:35 AM
09-23-2010 07:03 AM
make the following changes and test.
On the CME router,
conf t
voice service voip
h323
call start fast
On the h323 gateway page in CUCM,
Uncheck "Wait for far end terminal capability set"
Check "Enable Inbound Fast Start"
Reset the gateway in CUCM
-Felipe
09-23-2010 07:42 AM
I am afraid that this didn't work, as I have the same performance.
voice service voip
no notify redirect ip2ip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
h323
h225 h245-address on-connect
no call service stop
sip
session transport tcp
registrar server expires max 1200 min 300
midcall-signaling passthru
I am not sure why to change the settings on the gateway (even I have done it), since this does not work even internally.
09-24-2010 11:29 PM
Hello Felipe,
Do you have any further suggestions on this? Please let me know if you need additional logs.
kind regards,
George
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