SIP-UA Outgoing issue Cisco CME

[Kevin100]

Hai 

Our client CME SIP UA,  incoming calls are working  and out going calls not working.

If i restart router cme it will be  working  for 10 min.But SIP UA is registered always.

Main error message is Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41) for ccapi debug

and Reason: SIP ;cause=500 ;text="Classification Failure" for debug ccsip messages

please find attached debugs

[b.winter]

Hi,

where is the outgoing message sent to? To your SIP provider?

If yes, then the provider has to tell, what's wrong with the message.

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[Kevin100]

Hai 

Thanks for reply. This 2911 Cisco cme sip-ua router out going not working.Busy tone while dialing.

Here is the message ,debug ccsip messages

its from the router itself

 ITSP said they didn't receive any logs for out going.

Jan 4 06:31:52.856: //23/D6376C1C8031/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:65139333@alwatan.siptrk.kw:5060 SIP/2.0
Via: SIP/2.0/UDP 62.150.254.82:5060;branch=z9hG4bK12763
From: <sip:22247145@62.150.254.82>;tag=25FCA4-17FF
To: <sip:65139333@xxxxxxxx>
Date: Tue, 04 Jan 2022 06:31:52 gmt
Call-ID: D6D30FFD-6C5E11EC-803698DE-A5D8BC1D@xxxxxxxxx
Supported: 100rel,timer,resource-priority,replaces,histinfo
Min-SE: 1800
Cisco-Guid: 3593956380-1818104300-2150734046-2782444573
User-Agent: Cisco-SIPGateway/IOS-15.7.3.M8
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1641277912
Contact: <sip:22247145@xxxxxxxxx:5060>
History-Info: <sip:65139333@xxxxxxx:5060>;index=1
Expires: 60
Allow-Events: kpml, telephone-event
Session-ID: b8dae6a4dd48596ba70536e1b3262012;remote=00000000000000000000000000000000
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 385

v=0
o=CiscoSystemsSIP-GW-UserAgent 8720 2403 IN IP4 62.150.254.82
s=SIP Call
c=IN IP4 xxxxxxxxxxx
t=0 0
m=audio 16394 RTP/AVP 0 8 18 116 101
c=IN IP4 xxxxxxxxxx
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20

Jan 4 06:31:52.868: //23/D6376C1C8031/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP xxxxxxx:5060;received=xxxxxxx;rport=5060;branch=z9hG4bK12763
From: <sip:22247145@xxxxxxxxx>;tag=25FCA4-17FF
To: <sip:65139333@xxxxxxxx>;tag=1c297138129
Call-ID: D6D30FFD-6C5E11EC-803698DE-A5D8BC1D@62.150.254.82
CSeq: 101 INVITE
Reason: SIP ;cause=500 ;text="Classification Failure"
Content-Length: 0

Jan 4 06:31:52.872: //23/D6376C1C8031/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:65139333@xxxxxxx:5060 SIP/2.0
Via: SIP/2.0/UDP xxxxxxx:5060;branch=z9hG4bK12763
From: <sip:22247145@xxxxxxx>;tag=25FCA4-17FF
To: <sip:65139333@xxxxxxxx>;tag=1c297138129
Date: Tue, 04 Jan 2022 06:31:52 gmt
Call-ID: D6D30FFD-6C5E11EC-803698DE-A5D8BC1D@62.150.254.82
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: kpml, telephone-event
Session-ID: b8dae6a4dd48596ba70536e1b3262012;remote=cf705e5610b5509f9eda705525b1480a
Content-Length: 0

[b.winter]

Then you should take a packet capture on CUBE and check with Wireshark, where the CUBE sends the SIP packets to.

Either the CUBE sends the outgoing INVITE to somewhere wrong, or the provider is saying bull**bleep**.

One advice:

As you are using UDP, I would recommend configuring "connection-reuse via-port" in your tenant or global config.

Because every provider I have worked with, wants, that the CUBE sends the INVITE from the same source port, which the CUBE used for the initial REGISTER.

[Kevin100]

Hai

Thanks for the reply

If i do debug ccsip calls i am getting this message , do i need PVDM and transcoding in cme ? calls to ITSP using sip-ua , but initially after restart 10 min all calls will work.its strange.

Jan 4 08:29:21.626: //120/3F9596AC813B/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3EE23988
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 22247145
Called Number : 65139333
Source IP Address (Sig 62.150.254.82
Destn SIP Req Addr:Port : 62.150.105.203:5060
Destn SIP Resp Addr:Port : 62.150.105.203:5060
Destination Name : alwatan.siptrk.kw

Jan 4 08:29:21.626: //120/3F9596AC813B/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 62.150.254.82
Source IP Port (Media): 16476
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0

Jan 4 08:29:21.626: //120/3F9596AC813B/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 41
Disconnect Cause (SIP) : 500

[b.winter]

With the logs provided, I don't think it's a codec problem, so no need for PVDM/transcoding.

Reason code 41 is "temporary failure", which could be anything.

First, you should clarify, why the provider says, he doesn't see your INVITEs.

Then, they have to tell you, what is wrong with the INVITEs.

I would also suggest, that you restart the router again, so that the command I have provided above comes into effect.

And then take a trace from a working call and compare it with a none working call.

As I said in my original reply: As long as the provider doesn't say, what is wrong about the INVITE, you only can guess. Since they are the ones, who reply with an error message. You can't really do anything else.

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[Kevin100]

Dear B,Winter

Thanks for the reply , i check with provider and they inform me they didn't have any logs and calls not reaching them. Every time after restart everything will be ok for 10 min.There is some strange sip error comes inbetween,dont know it bug.attached same 

[Roger Kallberg]

Can you please share the current configuration of your gateway?

[Kevin100]

Dear Roger

Here is the config. Kindly note all working good for 10 min after restart.ITSP not get any logs after its not working.2911 latest ios

[b.winter · Accepted solution]

Hi,

I would recommend that you use tenants, because it's the "current standard" to configure CUBE.

And also, it's a must, if you have SRST running on the same router. I don't know, if it is a must for CME too, but I could imagine that.

The config could look something like this:

voice class tenant 100
 connection-reuse via-port --> Must be configured before the "credentials..." command"
 registrar dns:xxxxx:5060 expires 3600
 credentials username xxxxx password xxxxx realm xxxxx
 authentication username xxxxx password xxxxx realm xxxxxx
 nat symmetric check-media-src
 no remote-party-id
 retry invite 4
 retry response 3
 retry bye 2
 retry cancel 2
 retry register 5
 timers register 250
 sip-server dns:xxxxx:5060
 bind control source-interface GigabitEthernet0/1
 bind media source-interface GigabitEthernet0/1
 no remote-party-id
!
dial-peer voice 100 voip
 voice-class sip tenant 100
 no voice-class sip bind control source-interface GigabitEthernet0/1
 no voice-class sip bind media source-interface GigabitEthernet0/1
!
dial-peer voice 103 voip
 voice-class sip tenant 100
 no voice-class sip bind control source-interface GigabitEthernet0/1
 no voice-class sip bind media source-interface GigabitEthernet0/1
!
no dial-peer voice 101
no dial-peer voice 102
no sip-ua

Everything that you have in SIP-UA can be put into the tenant, which is then binded to the dial-peers towards SIP-provider.

And as your dial-peers 101 and 102 are already redundant to dial-peer 103, you can delete them (less code, easier to read).

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[Roger Kallberg · Accepted solution]

Instead of no sip-ua I would recommend these settings for the CME SIP phones to operate properly.

sip-ua 
 no remote-party-id
 retry invite 2
 timers trying 300
 registrar ipv4:<IP of the gateway internal int> expires 3600
 connection-reuse
 g729-annexb override

And for the inbound dial peer I would recommend this apart from what you already suggested.

voice class uri PSTN sip
host ipv4:<IP for SP SBC>
!
dial-peer voice 100 voip
 description Inbound calls from PSTN
 no session target registrar
 incoming uri via PSTN
voice-class sip tenant 100
dtmf-relay rtp-nte 

[Roger Kallberg]

Since you're using CME with SIP phones and they require a registrar I would advice you to configure the connection to your service provider with a tenant configuration instead of under SIP-UA. With this you can use SIP-UA registrar configuration for the CME SIP phones and use the tenant configuration on the applicable dial-peers for the connection to/from your service provider.

Apart from this you're dial peer configuration looks a bit messy as you have multiple similar or identical dial peers configured. Is that for any specific reason, or just happened to be so due to troubleshooting efforts? For one I think that you should not need to use both session sip-server and session registrar, one of these should be enough and you should try to clearly define one dial peer for inbound, for this it is advisable to use information in the VIA header to make the match, and then one or if needed more dial peer(s) for the outbound direction that has a distinct match for the destination pattern. Using dots like you do is not advisable as that is a to wide of a match.

Please see this excellent document on details for how to configure a tenant for SIP registration and also in general on how to configure dial peers. https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In-Depth-Explanation-of-Cisco-IOS-and-IO.html

[Kevin100]

Hai B.Winter/Roger

Thanks for reply and suggestions.

I will visit client and configure tenant .I have a mix of cme and sip phones registrar and sip-ua registrar. Its little mess

Dial peer redundant i shutdown not used. Just for testing configured.

[TONY SMITH]

You say that outbound calls work for around 10 minutes after you reboot the router.  If that's the case, then can you save SIP debugs for a good call straight after the router is restarted, and then for a bad call once it stops working?

In your second post you show an Invite which receives a 500 error. If the service provider isn't receiving your call attempt, where is that error 500 message coming from?

[Kevin100]

Dear Roger

Thanks for reply.I do have dns ,  router itself and i just removed and posted.

After tenant configuration incoming and out going OK,but sip phones got issues and i used your second config , then its fine. The invite 500 error was from the router itself and not from ITSP.

Thanks for all support