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New Member

3905 and SIP Trunk to PSTN: no ringback tone and no call processed

Hi,

We have a lot of Cisco 3905 IP Phones (fw 9-2-2ES3), and a cluster of CCM 8.6(2). To access PSTN we have a SIP Trunk (g729 codec).

The issue is that the 3905 can´t call to PSTN, they don´t receive ringback tone and the call is not processed. This issue doesnt apply to SCCP phones.

We have solved partially the problem, enabling MTP on the 3905 device configuration. With this feature enabled the calls to PSTN are normally processed.

I think this is no good, because everytime a 3905 phone makes a call (including interal calls) it will require MTP from the servers. And we´re going to have 4.000 of this phones on the cluster.

The other option is to enable MTP on the SIP Trunk, but in that case, every call to/from PSTN will require MTP from the servers, including those to SCCP phones.

There is something that I can required to SIP Trunk provider?

There is a way that the 3905 will dinamically obtain MTP only on PSTN calls? Mayber Early offer?

TIA

Andres Pasten

ENTEL S.A.

13 REPLIES
VIP Super Bronze

3905 and SIP Trunk to PSTN: no ringback tone and no call process

The first question is to know why the 3905 requires an MTP to make a call. Do you have a direct SIP trunk connection to your ITSP or do you have a gateway in between them? We will need to look either cucm traces of cube gateway debugs to know why the phones require MTP...There can be several reasons..DTMF is one...ut unless we look at logs we cant tell for sure

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New Member

Re: 3905 and SIP Trunk to PSTN: no ringback tone and no call pro

Hi,

Thanks for your sooner answer. There is no CUBE present on this solution. The SIP Trunk is configured directed to the CCM.

I've a trace attached now. If you search for the call generated from 227985, that´s a call processed to PSTN. If you search for the call generated from 227984, that´s a call with no ringback tone.

TIA

Andres Pasten

ENTEL S.A.

VIP Super Bronze

3905 and SIP Trunk to PSTN: no ringback tone and no call process

For the call to the PSTN, 227985, I dont see any INVITE going out..All I see is Notify, subscribe etc. Looks like the log is incomplete..

Please send me a detailed log and include the called number.

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Cisco Employee

3905 and SIP Trunk to PSTN: no ringback tone and no call process

HI Andres,

     Since you're running CUCM 8.6.2, on the SIP Profile of the Trunk you could enable "Early Offer if Required". This will insert an MTP on calls which require it only.

But, I agree with Aok, we need to find out why the 3905 phones require an MTP, please provide the Detailed CCM traces from all nodes in the cluster.

Regards,

Jagpreet

New Member

Re: 3905 and SIP Trunk to PSTN: no ringback tone and no call pro

Hi,

Thanks both of you for your support.

I've uploaded 2 traces, on both of them the called number is 23606902:

a) On trace from phone 227985, there is a call placed from a 3905 with MTP enabled, so the call goes correctly to the PSTN.

b) On trace from phone 227900, there is a call from 3905 with no MTP, so the call is not processed with no ringback tone.

I see a timer expired on the second trace, but I dont understand the cause.

TIA

Andres Pasten Calzada

ENTEL S.A.

Cisco Employee

3905 and SIP Trunk to PSTN: no ringback tone and no call process

Andres,

     The trace files which you attached do not have the complete call in them. I only see SIP messages ( Call processing and Stack ) and no other. Please follow these steps:

> Please make sure on the Service Ability Page >> Trace >> Configuration check all boxes just for testing purposes, we can revert them later.

> The trace level should be at Detailed.

> Make the test calls which you did and collect the traces from the RTMT if possible and not the CLI for the last 5-10 minutes.

Regards,

Jagpreet

New Member

Re: 3905 and SIP Trunk to PSTN: no ringback tone and no call pro

Jagpreet,

The CUCM and the SIP Trunk are on production, so I can only enable the first two requirements:

> Please make sure on the Service Ability Page >> Trace  >> Configuration check all boxes just for testing purposes, we can  revert them later.

> The trace level should be at Detailed.

But I can't isolate the traffic from other phones:

>  Make the test calls which you did and collect the traces from the RTMT  if possible and not the CLI for the last 5-10 minutes.

The traces provided are from only 2-3 minutes.

Please give some minutes to provide the new traces.

Andres

Cisco Employee

Re: 3905 and SIP Trunk to PSTN: no ringback tone and no call pro

Andres,

     No problem. Please take your time. you could always enable the traces after hours when no one is using the system or when the call traffic is low. I am attaching a screen shot of the settings to be enabled:

Regards,

Jagpreet

New Member

Re: 3905 and SIP Trunk to PSTN: no ringback tone and no call pro

Hi Jagpreet,

I've attached the requested traces.

Like in my previous post:

- Call from 227985, has MTP enabled, and proceed the call to PSTN

- Call from 227900, no MTP enabled, and call to PSTN is not placed

TIA

Andres Pasten

ENTEL S.A.

VIP Super Bronze

Re: 3905 and SIP Trunk to PSTN: no ringback tone and no call pro

Here is what I see in your traces. For the failed call, your provider never respond to your INVITE.

+++Here is an INVITE sent out to them, this was sent repeated;y but no response came back+++

8:19:39.370 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 10.254.134.97:[5060]:

[651145,NET]

INVITE sip:23606902@10.254.134.97:5060 SIP/2.0

Via: SIP/2.0/UDP 10.201.131.10:5060;branch=z9hG4bK1927e31ab0e

From: "Prueba Apasten" <223037900>;tag=223915~73c6d705-9b2e-48b2-8a69-10aa584b2502-60940237

To: <23606902>

Date: Tue, 23 Jul 2013 22:19:39 GMT

Call-ID: f34dbb00-1ef10177-dc-a83c90a@10.201.131.10

--

---

Content-Type: application/sdp

Content-Length: 239

v=0

o=CiscoSystemsCCM-SIP 223915 1 IN IP4 10.201.131.10

s=SIP Call

c=IN IP4 10.113.155.18

b=TIAS:8000

b=AS:8

t=0 0

m=audio 16390 RTP/AVP 18 101

a=rtpmap:18 G729/8000----------------------------------------AnnexB is not stated here, hence the default is yes

a=ptime:20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

In this INVITE I can see that you rae doing EO and in your SDP you have sent G729 with anexb set to yes.

From the trace of the working  call, we can see that your provider doesnt support annexb.

Here is the 200OK to the INVITE of the working call, the original invite was a DO (delayed offer)

+++

18:18:50.971 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 626 from 10.254.134.97:[5060]:

[650975,NET]

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.201.131.10:5060;branch=z9hG4bK18f5053edcc

From: <223037985>;tag=223859~73c6d705-9b2e-48b2-8a69-10aa584b2502-60940233

To: <23606902>;tag=SD12q2b99-d66xos2d-CC-31

Call-ID: d4e7bf80-1ef10144-db-a83c90a@10.201.131.10

CSeq: 101 INVITE

Contact: <23606902>

Content-Length: 196

Content-Type: application/sdp

v=0

o=HuaweiSoftX3000 350760 350761 IN IP4 10.254.134.97

s=Sip Call

c=IN IP4 10.254.134.97

t=0 0

m=audio 32856 RTP/AVP 18 8

a=rtpmap:18 G729/8000

a=rtpmap:8 PCMA/8000

a=fmtp:18 annexb=no--------------------------Annexb is not supported

Looking further in the trace, I can see that you have enabled EO on your sip trunk..

18:19:35.816 |//SIP/SIPCdpc(2,74,228)/ci=60940237/ccbId=223915/scbId=0/StartTransition: requireInactiveSDPForMidcallMediaChange=0, isTrunkEnabledForVoiceEO=1

18:19:35.816 |//SIP/SIPCdpc(2,74,228)/ci=60940237/ccbId=223915/scbId=0/sendPolicyAndRSVPRegisterReq: capCount[0], videoCap[0], dataCap[2], earlyOffer[2]

//SIP/SIPCdpc(2,74,228)/ci=60940237/ccbId=223915/scbId=0/outCall_waitRSVPRes_PolicyAndRSVPRegisterRes: policy[1], resvStatus[1], video[0] EOStatus[1]

I suggest you disable EO on your sip trunk and let your provider advertise what codecs they support and then CUCM will send an ACK based on that

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: 3905 and SIP Trunk to PSTN: no ringback tone and no call pro

Hi,

Thanks for your analisis, and suggestion, but customer detected that also transfers made from 3905 phones get dropped when the receptor of the transfer off-hook.

So I had to enable MTP on the SIP Trunk to correct this, nothing else worked.

I'll try your configuration later, when customer is out of the office. I´ll let you now wht is obtained.

TIA

Andres Pasten

ENTEL S.A.

New Member

Re: 3905 and SIP Trunk to PSTN: no ringback tone and no call pro

Hi,

I removed the EO from the Sip Trunk as you said, and the Cisco 3905 processed the call normally.

But then I remembered why the EO was configured, because the conference needed it to function.

So I'm trapped. I can't remove the MTP from the 3905 if I want to have conferences.

If you have any ideas, please let me know.

Andres Pasten

ENTEL S.A.

VIP Super Bronze

Re:3905 and SIP Trunk to PSTN: no ringback tone and no call proc

What type of conference solution are you using and why do you need EO for it.


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