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Community Member

Sip calls issue via H323

Hi,

I have the follow environment:

ip phones (SCCP and SIP) ----------- CME-SRST(Site B) --------H323-----------CUCM (Site A)

                                                        |

                                                   PSTN

Every phones are registered on CUCM (site A) in SRST mode.

The SCCP phone are doing outbound call normally via PSTN local, but the SIP phones calls are dropping.

thanks. 

Everyone's tags (4)
18 REPLIES
Cisco Employee

Sip calls issue via H323

Are you saying that this problem happens only in SRST mode?

Can you post the configuration for the Site B CME-SRST?

Community Member

Re: Sip calls issue via H323

Yes, befored SRST configuration the environment was working with SIP phones registered on CME. 

I have in the site A SIP phones (registered on CUCM) and it can to do outbound call via site B PSTN normally.

My problem is only with SIP phones of the site B that are registered on CUCM (site A). 

      

The run file was attached.

Cisco Employee

Sip calls issue via H323

Again, just to clarify:

Originally, you had two sites, A with CUCM and B with CUCME.  Both sites have SCCP and SIP phones registered to their local CUCM/CUCME.

Now, you've configured the Site B CUCME router for SRST and the Site B phones (SCCP and SIP) are registering to the CUCM at Site A.

In this new scenario, all site A phones (SCCP and SIP) can make outbound calls to the PSTN at Site B fine.  Also, the SCCP phones at Site B can make outbound calls to the PSTN.

The problem is with the SIP phones at Site B, which are registered to the CUCM at Site A.  They can not make outbound calls to the PSTN.


Is this correct?

If so, based on your last attached debug, I'm suspecting a codec issue.  Are your media resourses and MRGL configured properly?

Community Member

Re: Sip calls issue via H323

Correct! Gregory.

I need to create a Xcoder between CUCM and CME-SRST?

I didn´t any configuration with MRGL after srst configuration.

My SRST-CME (172.23.0.1) router has:

dspfarm profile 2 transcode

codec g729abr8

codec g729ar8

codec g711alaw

codec g711ulaw

codec g729r8

codec g729br8

maximum sessions 2

associate application SCCP

!

dspfarm profile 1 conference

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 1

associate application SCCP

it7-rot-df-01#  sh sccp
SCCP Admin State: UP
Gateway Local Interface: Vlan1
        IPv4 Address: 172.23.0.1
        Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 172.23.0.1, Port Number: 2000
                Priority: N/A, Version: 7.0, Identifier: 10
                Trustpoint: N/A

Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 172.23.0.1, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 2
Reported Max Streams: 4, Reported Max OOS Streams: 0
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: g729br8, Maximum Packetization Period: 60
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30

Conferencing Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 172.23.0.1, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1
Reported Max Streams: 8, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: g729br8, Maximum Packetization Period: 60
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30

thanks.

Cisco Employee

Sip calls issue via H323

It depends.  Since you have two sites, CUCM may have the devices is different regions. All of the SIP configuration on the CME doesn't come into play if the phones are registering to CUCM.

Look at your media resources in CUCM to see if they're registering correctly.  Check your region settings, specifically what codecs are being used between regions.  Also, check to see how the SIP phones are configured compared to a working SCCP phone from Site B.  For instance, do they share the same DP, CSS, etc.

Community Member

Re: Sip calls issue via H323

I have the devices in different regions, device pool, PT and CSS...

I´m using codec default g711 between the sites and the media resources are configured normally and registered.

The gateway configuration is with site A device pool and CSS.

To change de device pool and css of the gateway config is a good test?

Cisco Employee

Re: Sip calls issue via H323

Before you do that, check to see how the SIP phones are configured compared to a working SCCP phone from Site B.  For instance, do they share the same DP, CSS, etc.

Community Member

Re: Sip calls issue via H323

right, it is ok!

Cisco Employee

Re: Sip calls issue via H323

By that do you mean that at Site B you have a SIP and an SCCP phone, both with identical PT, CSS, DP, etc.?

And, with that identical setup, only the SIP phones are having problems?

Community Member

Re: Sip calls issue via H323

Yes, correct. The site B have SIP and SCCP phones and both are with the same PT, CSS, DP... But only sip phones are having problem.

Community Member

Re: Sip calls issue via H323

I did a test now and "sh voice call status" show the call, but it is dropped.

it7-rot-df-01#sh voice call status

CallID     CID  ccVdb      Port        Slot/DSP:Ch  Called #   Codec    MLPP Dia                                                                                        l-peers

0x1C224    33FB 0x312EFB90 0/1/2            0/1:1  *084934444  None     0/104

Cisco Employee

Sip calls issue via H323

Can the SIP phones at Site B make calls to internal extensions?  Can they recieve calls from outside the network?

Community Member

Re: Sip calls issue via H323

Yes. The sip phones at site B make calls to internal extensions.

My router is configured with "connection plar" forwarding the call for SCCP phone. It is working properly.

Community Member

Re: Sip calls issue via H323

Gregory,

The SIP phones directory number does not have a dial-peer how we can to see bellow.

I´m going to create a dial-peer voip for them. What do you think?

it7-rot-df-01#sh dial-peer voice summary

dial-peer hunt 0

             AD                                    PRE PASS                OUT

TAG    TYPE  MIN  OPER PREFIX    DEST-PATTERN      FER THRU SESS-TARGET    STAT PORT    KEEPALIVE

100    voip  up   up             70[0-4].$          0  syst ipv4:172.20.1.106

101    voip  up   up             70[7-9].$          0  syst ipv4:172.20.1.106

102    pots  up   up             0[2-5].......$     0                      up   trunkgroup FXO

103    voip  up   up             0041........       0  syst ipv4:192.168.10.3

104    pots  up   up             0[6-9].......$     0                      up   trunkgroup FXO

105    voip  up   up             705.$              0  syst ipv4:172.22.0.1

106    pots  up   up             00[1-3]T           0                      up   trunkgroup FXO

107    pots  up   up             004[2-9]T          0                      up   trunkgroup FXO

108    pots  up   up             00[5-9]T           0                      up   trunkgroup FXO

109    pots  up   up             010T               0                      up   trunkgroup FXO

110    pots  up   up             01[1-9].           0                      up   trunkgroup FXO

111    pots  up   up             010[1-9]..         0                      up   trunkgroup FXO

             AD                                    PRE PASS                OUT

TAG    TYPE  MIN  OPER PREFIX    DEST-PATTERN      FER THRU SESS-TARGET    STAT PORT    KEEPALIVE

20001  pots  up   up             7060$              0                           50/0/1

20002  pots  up   up             7061$              0                           50/0/2

20003  pots  up   up             9000$              0                           50/0/20

20004  pots  up   up             9000$              0                           50/0/21

20005  pots  up   up             9000$              0                           50/0/22

20006  pots  up   up             9000$              0                           50/0/23

20007  pots  up   up             9000$              0                           50/0/24

20008  pots  up   up             7065$              0                           50/0/25

7060   voip  up   up             7060               0  syst ipv4:172.20.1.106

Community Member

Re: Sip calls issue via H323

Hi Gragory!!!

The SIP phones are making outbound calls now!!!

The dial-peer voip was the solution.

thanks.

Cisco Employee

Re: Sip calls issue via H323

Good to hear!

Can you elaborate on the solution?

Community Member

Re: Sip calls issue via H323

I put the follow:

dial-peer voice 706 voip

corlist outgoing DF

description LIGAR-CURITIBA-1

destination-pattern 706.$

session target ipv4:172.20.1.106

voice-class codec 1

voice-class h323 1

dtmf-relay h245-alphanumeric

no vad

At the moment of the call o codec is being negotiated right now.

thanks.

Cisco Employee

Re: Sip calls issue via H323

You wouldn't need any dial peers for the directory numbers on the SIP phones.

It doesn't look like you have an MTP setup on the gateway (CME). 

On the gateway configuration for the CME router in CUCM do you have the "Require MTP" box checked off?

You may need to create the MTP on the gateway and register it to CUCM for it to work.

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