05-15-2012 01:23 PM - edited 03-16-2019 11:10 AM
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.
05-15-2012 01:55 PM
Are you saying that this problem happens only in SRST mode?
Can you post the configuration for the Site B CME-SRST?
05-16-2012 05:41 AM
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.
05-16-2012 07:26 AM
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?
05-16-2012 08:03 AM
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.
05-16-2012 08:47 AM
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.
05-16-2012 10:11 AM
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?
05-16-2012 10:14 AM
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.
05-16-2012 10:15 AM
right, it is ok!
05-16-2012 10:17 AM
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?
05-16-2012 10:20 AM
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.
05-16-2012 10:18 AM
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
05-16-2012 10:27 AM
Can the SIP phones at Site B make calls to internal extensions? Can they recieve calls from outside the network?
05-16-2012 10:32 AM
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.
05-16-2012 10:38 AM
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
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