beep tone via outgoing SIP trunk 3

[boygie10]

Need your help guys

When do outbound via SIP trunks, we hear beep tone pause beep tone pause instead of a normal ringing.

We can still able to connect to the end party with no issue but the only problem is the ringing tone outbound.It sounds like an international ring rather than a normal ring where calls are made locally or within the country only.

anything to adjust on CM or SM or SBC?

the SIP provider is Verizon

 

[montyzummer]

If its your carrier providing the tones its up to them, this sounds like its vorizon ring back to me.

You can change tone gen "change tone 1- 25" but i would step away from the tone , there is nothing to see here move along please sir.

Also multifrequence signalling can be altered.

There are some verizon specific settings in special apps but these are header settings , rather that ringback.

So on your head be it.

ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.

 

[SureshGanti]

I would not suggest to change the tones on the System as they are sytem wide parameters. This is happening only for that specific SIP trunk, then its more localized to that SIP Trunk and that provider.

On Normal telephone lines, it's the connected exchange who provides the ring back tone. If you are connected with Verizon, its verizon who provides the ringback tone. This is normal for PSTN trunks, not sure of SIP trunks though.

I would still suggest to please review the SIP Trunk settings, specifically the country parameters and coding, also check with Verizon if they can provide specific ring back tone that you are looking for.

Thanks
Suresh Ganti

 

[kyle555]

I politely disagree with Monty on this one! I suggest you go check what's in those tone plans. Or, at the very least make yourself a VDN/vector internally that "plays ringback for 5 seconds" to see what it does.

On a SIP flow where you INVITE and the far end is 100 trying then 180 ringing, an RTP stream usually hasn't been set up yet. I don't give telco RTP ringback when I get called between my 180 ringing and 200 OK, nor do they when I call out through them.
Analog/digital/h323 should get that from the PBX, SIP sets probably play it themselves. What type and firmware on your phones?

 

[boygie10]

@kyle555- below is the firmware


Station Ext Set Type/ Prod ID/ TCP Station IP Address/
or Orig Port Net Rgn Release Skt Gatekeeper IP Address
------------- --------- ---------- --- ---------------------------------------
4405621 9620 IP_Phone y 10.243.58.76
13 3.250A 10.245.52.23
Appreciate where I can check this.
'On a SIP flow where you INVITE and the far end is 100 trying then 180 ringing, an RTP stream usually hasn't been set up yet. I don't give telco RTP ringback when I get called between my 180 ringing and 200 OK, nor do they when I call out through them'

Issue still exist and also hearing the same ringback from incoming calls going to h323 extension using outbound/inbound SIP trunks.

 

[kyle555]

3.250. Either way, it's H323. If you did a list trace on that station calling out the SIP trunk, you should see dialing, picking a trunk, a media path from the gateway to the set for ringback, and upon answer, a media path from the SBC to the gateway as well as the gateway to the set.

Now, I haven't tested it, but in your sig group out to the sip trunks, you have the option of early h323 station media. It's going to make CM send the phone's IP as the first place the SBC/SIP carrier should send audio and not go to a gateway first and then send a reinvite with new SDP right to the set to shuffle after it's answered. That's how I understand it affects the far-end side of the signaling. What that does between CM and the H323 phone, I don't know. Maybe the phone gets ringback from a gateway and the phone shuffles to the SBC, but the SBC never has to shuffle to the phone?
Or, maybe that parameter might make the H323 phone play local ringback to itself.

In either case, with a solid trace you should know where that media is coming from.

 

[boygie10]

Hi Kyle555

I have an outbound trace below hitting a SIP trunk; however, not sure where is the 'media path' here.

I have also checked the sig group- but afraid cant see the ' option of early h323 station media'??

You been very helpful Kyle and do hope you stay to support.

Thanks

Pedro

me data

15:39:33 TRACE STARTED 03/27/2017 CM Release String cold-03.0.124.0-22147
15:39:33 G711A ssff ps:20
rgn:7 [10.243.226.29]:2618
rgn:7 [10.243.224.36]:2058
15:39:39 dial 907399355667 route:ARS
15:39:39 term trunk-group 780 cid 0x19f9
15:39:39 dial 907399355667 route:ARS
15:39:39 route-pattern 780 preference 1 location 13 cid 0x19f9
15:39:39 seize trunk-group 780 member 10 cid 0x19f9
15:39:39 Calling Number & Name NO-CPNumber NO-CPName
15:39:39 SIP>INVITE sip:00447399355667@eu.jllnet.com SIP/2.0
15:39:39 Call-ID: 802c20ebdefe71805558da293200
15:39:39 Setup digits 00447399355667
15:39:39 Calling Number & Name NO-CPNumber Guanco, Pedro
15:39:39 Proceed trunk-group 780 member 10 cid 0x19f9

time data
15:39:40 G711A ssff ps:20
rgn:250 [10.228.64.43]:20352
rgn:7 [10.243.224.36]:2064
15:39:40 xoip options: fax:T38 modemff tty:US uid:0x50142
xoip ip: [10.243.224.36]:2064
VOIP data from: [10.243.224.36]:2064
15:39:44 Jitter:2 0 0 0 0 0 0 0 0 0: Buff:19 WC:13 Avg:2
15:39:44 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
15:39:45 Alert trunk-group 780 member 10 cid 0x19f9
VOIP data from: [10.243.224.36]:2058
15:39:45 Jitter:0 0 0 0 0 0 0 0 0 0: Buff:11 WC:0 Avg:0
15:39:45 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
VOIP data from: [10.243.224.36]:2064
15:39:53 Jitter:2 1 1 2 2 3 4 2 2 0: Buff:24 WC:16 Avg:2
15:39:53 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
VOIP data from: [10.243.224.36]:2058
LIST TRACE

time data
15:39:53 Jitter:0 0 0 0 0 0 0 0 0 0: Buff:11 WC:0 Avg:0
15:39:53 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
15:39:57 SIP>ACK sip:447399355667@10.228.64.43:5060;transport=tcp;gs
15:39:57 SIP>id=802c20eb-de0f-4701-bd55-58da29320000 SIP/2.0
15:39:57 Call-ID: 802c20ebdefe71805558da293200
15:39:57 idle trunk-group 780 member 10 cid 0x19f9
15:39:58 idle station 4436516 cid 0x19f9
15:39:58 G711A ssff ps:20
rgn:7 [10.243.226.29]:2618
rgn:7 [10.243.224.36]:2068
15:39:58 idle station 4436516 cid 0x1a16
15:40:01 TRACE COMPLETE station 4436516 cid 0x0

 

[kyle555]

Well, it looks like your speech path is set up from the moment you finish dialing - in the g711A lines with IP info. So, at the top is you getting dialtone.

Is .226.29 your set and .224.36 the gateway? What's the SBC IP? If you enable early h323 station media in the sig group to that trunk and did a traceSM/wireshark on the station, you could see if that's coming back from the carrier as soon as you finish dialing.

If you're on a Avaya SBC, it's easy enough in the webpage to set a packet capture on the inside interface and download a PCAP to see it without having to port mirror on the LAN for the phone.

 

[boygie10]

yes.226.29 is station ip and .224.36 is the MG.

I'm not quite sure really where to find or enable the 'early h323 station media'- i was in sig group parameters- but cant see one?

 

[boygie10]

@kyle555

I have just found the parameters to enable station direct media-- is this not a service affecting command?

Enforce SIPS URI for SRTP? y
Peer Detection Enabled? y Peer Server: SM
Prepend '+' to Outgoing Calling/Alerting/Diverting/Connected Public Numbers? y
Remove '+' from Incoming Called/Calling/Alerting/Diverting/Connected Numbers? n
Alert Incoming SIP Crisis Calls? n
Near-end Node Name: procr Far-end Node Name: UKLONVASE001
Near-end Listen Port: 5061 Far-end Listen Port: 5061
Far-end Network Region: 250

Far-end Domain: eu.jllnet.com
Bypass If IP Threshold Exceeded? n
Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n
DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y
Session Establishment Timer(min): 3 IP Audio Hairpinning? n
Enable Layer 3 Test? y Initial IP-IP Direct Media? n
H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6

 

[montyzummer]

Hey Kyle , thankyou for being polite ,

ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.