SIP trunk early offer

Unanswered Question
Jun 14th, 2012

Hi,

We have configured a trunk with a provider using CUBE.

Callmanager--------CUBE--------Provider

The provider wants early offer and G729r8

So we configured a SIP trunk with a device pool/region so that only g729 is allowed between SIP trunk and the rest.

We have configured an IOS MTP resource, and this is registered on callmanager:

!

dspfarm profile 2 mtp

codec g729r8

maximum sessions software 20

associate application SCCP

!

!

The mtp resource is assign to the trunk using MR-list and MR-group

The trunk has MTP enabled with "MTP prefered Codec" G729b/G729ab

On CUBE we enable "deb ccsip mess" and we see the invite comming from callmanager, but without attached SDP

What must be done to make callmanager use early-offer?

Thanks for the help,

Jan

I have this problem too.
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Joe Martini Thu, 06/14/2012 - 15:01

Check MTP required on the SIP trunk configuration page or if you are on CUCM 8.5 or later go to the SIP Profile (Device > Device Settings > SIP Profile) and check "Early Offer support for voice and video calls (insert MTP if needed)".

j.huizinga Thu, 06/14/2012 - 22:23

Hello Joe,

I have MTP required enabled on the trunk, but it doesn't really change anything.

If I enable, or don't enable "MTP required" the SIP INVITE from callmanager is the same.

We are running Callmanager 7.1.5, so we don't have the option in the SIP profile.

When we establish a call I can see that the IOS MTP is involved (show sccp connections) but no early offer.

To make a successfull cal we force the provider dial-peer to use G729 only.

Unfortunately this IOS doesn't have the forced early offer option.

Is there someone who has succeeded in making CM 7.1.5 do an early offer?

Thanks,

Jan

nikshah Fri, 06/15/2012 - 00:07

Hi Jan,

Here is your answer

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/media.html#wp1055833

Snippet from above

SIP Early Offer

SIP negotiates media exchange via Session Description Protocol (SDP),  where one side offers a set of capabilities to which the other side  answers, thus converging on a set of media characteristics. SIP allows  the initial offer to be sent either by the caller in the initial INVITE  message or, if the caller chooses not to, the called party can send the  initial offer in the first reliable response. By default, Unified CM  sends the INVITE without an initial offer, and it requires MTP resources  to send the offer in the INVITE. Note that this initial offer is  limited to the G.711 codec only.

Also note that MTP resources are not required for incoming INVITE messages, whether or not they contain an initial offer.

Regards

Niket

j.huizinga Fri, 06/15/2012 - 04:06

Hi Niket,

You seem to be right, when I force 711 I see early offer.

But it is weird that on the trunk configuration page, it allows me to select a g729 codec as prefered codec.

Can you explain why this option is there, although it doesn't sem to work?

Thanks for your help,

Jan

nikshah Fri, 06/15/2012 - 14:34

Hi Jan,

I dont know whats the logic and i dont have a confirmed answer, behind not supporting the g729 codec in CM 7.1.5 , however when i open the SRND for CM 8.x

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/trunks.html#wp1126420

I dont see any requirements on the codec.

So somewhere between CM 7.x and CM 8.x there must be a fix for supporting the g729 codec on early offer.

Hope this helps.

Regards

Niket

j.huizinga Sun, 06/17/2012 - 11:56

Hi Niket,

Thank you very much.

We are testing now with another router that has an IOS which supports 'SIP early offer forced' and we hope that this will work.

So the callmanager will send a normal INVITE (delayed offer) and the CUBE will send out a early offer to the provider.

Regards,

Jan

Ayodeji oladipo... Mon, 06/18/2012 - 00:51

J,

Did the early offer-forced work?

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j.huizinga Mon, 06/18/2012 - 03:10

Hi,

We haven't configured yet. I shall let the community know.

Thanks for all,

Jan

Ayodeji oladipo... Tue, 06/19/2012 - 00:26

J,

I just saw your query on the ask the expert forum. I thought I should chip in here for you.

Yes CUCM can send delay offer and CUBE can send early offer. This is how we have it configured for one of my biggest customers with 3 clusters (8 server/cluster) and it works very well...

Here is a snippet from our CUBE config..

sip

early-offer forced.

Here is a sample trace..

+++++CUCM sends invite without early offer+++++++++

Received:
INVITE sip:07544455678@172.16.10.74:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.174:5060;branch=z9hG4bKa41e64e002602
From: "Chelmsford" ;tag=517285~ffa80926-5fac-4dd6-b405-2dbbc56ae9a2-400694295
To:
Date: Mon, 18 Jun 2012 09:46:00 GMT
Call-ID: 67d44180-fde1f8d8-3aead-ae28690a@172.16.10.174
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 1741963648-0000065536-0000064326-2921883914
Session-Expires:  84600
Contact:
Max-Forwards: 70
Content-Length: 0

++++++++++++++CUBE sends a trying to CUCM++++++++++++++++

Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.10.174:5060;branch=z9hG4bKa41e64e002602
From: "Chelmsford" ;tag=517285~ffa80926-5fac-4dd6-b405-2dbbc56ae9a2-400694295
To:
Date: Mon, 18 Jun 2012 09:46:00 GMT
Call-ID: 67d44180-fde1f8d8-3aead-ae28690a@172.16.10.174
CSeq: 101 INVITE
Allow-Events: kpml, telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

+++++++++CUBE sends an invite to ITSP with Early Offer++++++++++


010625: Jun 18 09:46:00.371: //452987/67D441800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:07544455678@10.100.33.54:5070 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.74:5060;branch=z9hG4bKFE71F2421
Remote-Party-ID: "Chelmsford" ;party=calling;screen=no;privacy=off
From: "Chelmsford" ;tag=88EFEFBA-14A2
To:
Date: Mon, 18 Jun 2012 09:46:00 GMT
Call-ID: 3EDB1C4A-B86111E1-94E48F4D-5D7E5E41@172.16.10.74
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1741963648-0000065536-0000064326-2921883914
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1340012760
Contact:
Expires: 180
Allow-Events: kpml, telephone-event
Max-Forwards: 69
Session-Expires:  84600
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 354

v=0
o=CiscoSystemsSIP-GW-UserAgent 6030 604 IN IP4 172.16.10.74
s=SIP Call
c=IN IP4 172.16.10.74
t=0 0
m=audio 18110 RTP/AVP 18 0 8 100 101
c=IN IP4 172.16.10.74
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

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j.huizinga Tue, 06/19/2012 - 00:46

Thank you very much!

We are going to configure it like this.

Unfortunately one of our CUBE router has not the forced SIP early offer command, and we need to install more flash and memory and then do an IOS upgrade.

We now make a test with a 2811 that has the correct IOS, but this router has such a weird issue that I think I have to open a TAC case. In this router I can not apply the "bind control source-interface" command. I can type it, but the"show run" doesn't show it. So weird.

We did format flash, install another IOS, but it doesn't apply the bind commands.

Thanks!

Jan

Ayodeji oladipo... Tue, 06/19/2012 - 00:52

How did you apply the command..Is it like this..

sip

  bind control source-interface gig0/0

  bind media source-interface gig0/0

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Ayodeji oladipo... Tue, 06/19/2012 - 00:56

What IOS version are you running?

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j.huizinga Tue, 06/19/2012 - 00:59

c2800nm-ipvoice_ivs-mz.124-20.T6.bin

c2800nm-adventerprisek9-mz.124-24.T7.bin

c2800nm-adventerprisek9-mz.124-24.T4.bin

c2800nm-adventerprisek9-mz.124-24.T5.bin

Pick one

We tried all of them.

Thanks,

Jan

Ayodeji oladipo... Tue, 06/19/2012 - 01:37

Can you send the output of this command..

show sip-ua status

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j.huizinga Tue, 06/19/2012 - 01:50

Hi,

At this moment I have removed this router from customer site, and it is in my office.

We got crazy about this router not accepting the bind commands.

But regardless of configuration, the bind commands should be accepted I think.

Thanks,

Jan

jfernandorosa Mon, 09/02/2013 - 11:41

Hi Aokanlawon,

I have the similar problem:

My scenario:

CUCM 9X -->SIP TRUNK--> CUBE --> ISP SIP

I need invite Early Offer to ISP, for DTMF problems, I don´t like the use CUCM fot this.

I set in CUBE (early-offer forced), but if I removed pass-thru content sdp, i received fas busy and CUCM return Internal Server Error:

sip

  bind control source-interface GigabitEthernet0/1

  bind media source-interface GigabitEthernet0/1

  early-offer forced

  midcall-signaling passthru

  pass-thru headers unsupp

  no call service stop

how can I solve this?

Thanks!

Joao

Ayodeji oladipo... Mon, 09/02/2013 - 12:36

Can you do a test call and send us "debug ccsip messages" attach it here

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Ayodeji oladipo... Mon, 09/02/2013 - 12:54

The log you attached was a succesful call and your cube didnt send EO to your ITSP. I didnt see any error in the log

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jfernandorosa Mon, 09/02/2013 - 13:41

In this case not have problem, is correct.

But in some calls the ISP don´t invite SDP payload with DTMF information (telephone event) and DTMF fails in this case.

I attached log problem.

Thanks.

Joao

Ayodeji oladipo... Mon, 09/02/2013 - 14:30

There is nothing you can do, if your ITSP doesnt advertise any DTMF capabilites in their SDP. You need to contact them and have it corrected. CUBE can only respond to what is offered. This is a problem with them so get them to sort it out

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jfernandorosa Mon, 09/02/2013 - 15:02

Hi.

I remove the SIP configs below for invite SDP EO to ISP.

sip

  pass-thru headers unsupp

  pass-thru content sdp

  no call service stop

and use (early-offer forced)

The ISP response with payload complete in this case, I dialed the same number with problems, but my call rinring and return a fast busy in this case, CUBE return internal server error for CUCM.

Sent:

SIP/2.0 500 Internal Server Error

Via: SIP/2.0/TCP 21.10.0.7:5060;branch=z9hG4bK13eb293705d2

From: "ATA187 Core" ;tag=21576~fb89236f-816b-47f5-8c94-b8d3c388dd7c-64665064

To: ;tag=3E21E738-DD8

Date: Mon, 02 Sep 2013 21:44:24 GMT

Call-ID: 263d9280-22510742-b70-7000a15@21.10.0.7

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.2.4.M3

Reason: Q.850;cause=96

Content-Length: 0



Ayodeji oladipo... Tue, 09/03/2013 - 01:45

You need to post the full debug, for us to know whats happening. Cause code 96 means that a madatory IE is missing

Typical scenarios include:

  • Mandatory Contact field missing in SIP message.
  • Session Description Protocol (SDP) body is missing.

So until I see the full log, I wont know what is wrong

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jfernandorosa Tue, 09/03/2013 - 07:00

Hi,

Attached logs with error 96 for you analise.

I don´t see the SDP payloads.

Thanks for help.

Joao

jbollen Tue, 09/03/2013 - 07:50

Well, in this case (error code 96) the service provider is:

a. removig the SDP in the subsequent 180 Ringing

b. not including an SDP in the 200 OK

That's why the router spits out cause code 96.

Talk to the Service Provider, I would suggest...

cheers,

Jan

Ayodeji oladipo... Tue, 09/03/2013 - 08:34

Hi,

I have looked at the traces and the problem is from your CUBE. Here is my analysis

1. When ITSP sent the first 180 ringing with SDP, it indicated that it wanted to do early media and requested a PRACK.

"Require: 100rel"

Received:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 21.10.0.3:5060;branch=z9hG4bKBA319D8

From: "ATA187 Core" ;tag=3E21E5CC-34E

To: ;tag=277bus7u-CC-42

Call-ID: AAE9B01E-134F11E3-B2E7A8D4-D9EFC3D@21.10.0.3

CSeq: 101 INVITE

Timestamp: 1378158264

Contact:

Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER

Require: 100rel

RSeq: 1

Content-Length: 228

Content-Type: application/sdp

v=0

o=HuaweiSoftX3000 12465003 12465003 IN IP4 10.57.0.117

s=Sip Call

c=IN IP4 10.57.0.117

t=0 0

m=audio 19728 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=fmtp:18 annexb=no

2. However when CUBE sent a PRACK to the ITSP it included no answer to their offer, ie no SDP.

Sent:

PRACK sip:40042484@10.56.14.18:5060;user=phone;transport=udp SIP/2.0

Via: SIP/2.0/UDP 21.10.0.3:5060;branch=z9hG4bKBA4169E

From: "ATA187 Core" ;tag=3E21E5CC-34E

To: ;tag=277bus7u-CC-42

Date: Mon, 02 Sep 2013 21:44:24 GMT

Call-ID: AAE9B01E-134F11E3-B2E7A8D4-D9EFC3D@21.10.0.3

CSeq: 102 PRACK

RAck: 1 101 INVITE

Allow-Events: telephone-event

Max-Forwards: 70

Content-Length: 0

As you can see there is no SDP in the PRACK sent to the ITSP. This is where everything broke. The next few lines, ITSP then sent another 180 ringing without SDP and still requested PRACK, perhaps hoping that CUBE will send offers in its PRACK, but CUBE still ddint send anything...

PRACK is used to establish early media or to cut through audio on the PROGRESS message or ringing in this case. Hence there is no way this can be achieved if CUBE doesnt send SDP in its PRACK message in this scenario.

So we need to find out why CUBE is behaving this way. Please attach a sh run of your gateway.

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jfernandorosa Tue, 09/03/2013 - 10:02

Hi,

I remove in sip-ua:

disable-early-media 180

The calls in work fine!

Thanks for help!

Regards.

Joao

jfernandorosa Tue, 09/03/2013 - 11:20

Hi,

After this change, my fax services don´t work, I call to number receive a  ring, receive a signal and return a fast busy.

I attached logs.

Joao

Ayodeji oladipo... Tue, 09/03/2013 - 11:35

I can see that the call starts as g729 and changed to G711 when a fax tone was detected at the far end..

What is the ip address of the device your fax is connected to? What is the region setting between your fax device and the gateway?

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jfernandorosa Tue, 09/03/2013 - 11:50

IP Endpoint ATA187 is: 21.10.1.50

Region between Fax and Gateway is G711

.

Thanks!

Joao

Ayodeji oladipo... Tue, 09/03/2013 - 12:05

You need to reconfigure the dial-peers for your fax devices. The voice call setup is sent to a xcoder..

From the logs, RTP stream is sent to ip address 21.10.0.2.

Received:

ACK sip:297832422526@21.10.0.3:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 21.10.0.7:5060;branch=z9hG4bK153f6bc5cebe

From: "ATA187 Core" ;tag=24382~fb89236f-816b-47f5-8c94-b8d3c388dd7c-64665213

To: ;tag=4272C5B4-D17

Date: Tue, 03 Sep 2013 17:53:30 GMT

Call-ID: bcc58e80-2261221a-c15-7000a15@21.10.0.7

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: presence, kpml

Content-Type: application/sdp

Content-Length: 251

v=0

o=CiscoSystemsCCM-SIP 24382 1 IN IP4 21.10.0.7

s=SIP Call

c=IN IP4 21.10.0.2

That suggest that the call is routed to a transcoder or MTP device...can you confirm what device is this.

If the region between your ATA and CUBE is G711, then you need to have the inbound dial-peer from CUCM advertise G711 and your outbound dial-peer to your ITSP using G711 for fax calls only, all other calls can use G729

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jfernandorosa Tue, 09/03/2013 - 12:12

Hi,

21.10.0.2 is a CUBE with MTP and transcoder.

I forced call to use G711 in dial-peer (in and out), but not successfully, attached logs with G711.

Thanks.

Joao

Ayodeji oladipo... Tue, 09/03/2013 - 12:28

I can see the call using G711 however rtp is still sent to c=IN IP4 21.10.0.6.

For your fax to work media/rtp stream has to be sent directly to the ATA.

Looking at your config, I can see that this device is a MTP device. So there is a DTMF mistmact hence calls are terminated on the MTP..

Try and add this to your inbound voip dial-peer

dial-peer voice 2 voip

description # Call Leg VOIP #

incoming called-number .

dtmf-relay rtp-nte digit-drop sip-kpml

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jfernandorosa Tue, 09/03/2013 - 12:35

21.10.0.6 is a CUCM.

I can´t insert the RTP comands, in inbound dial-peer is not exist:

ra076963.igs.cref(config-dial-peer)#dtmf-relay rtp-nte digit-drop ?

  cisco-rtp          Cisco Proprietary RTP

  h245-alphanumeric  DTMF Relay via H245 Alphanumeric IE

  h245-signal        DTMF Relay via H245 Signal IE

 

I insert:

dial-peer voice 2 voip

description # Call Leg VOIP #

incoming called-number .

dtmf-relay rtp-nte digit-drop

Not working.

Thanks!

Joao

Ayodeji oladipo... Tue, 09/03/2013 - 12:40

Dont you have a sip trunk between CUCM and CUBE? If you do then your dial-peer 2 should have session protocol sipv2 on it. You need the dial-peer to be enabled for sip to add that command

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jfernandorosa Tue, 09/03/2013 - 12:56

Yes, I have the SIP Trunk between CUCM and CUBE.

I change the dial-peer to:

dial-peer voice 2 voip

description # Call Leg VOIP #

session protocol sipv2

incoming called-number .

voice-class codec 1

dtmf-relay rtp-nte digit-drop sip-kpml

But not working.

Attached logs.

Thanks!

Joao.

Ayodeji oladipo... Tue, 09/03/2013 - 13:08

The call is still sent to MTP not the fax device. What do you have configured on your sip trunk dtmf method? Is it rfc2833 or no preference?

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Ayodeji oladipo... Tue, 09/03/2013 - 13:14

ok..Can you go to the ATA187 config on cucm and check if you have the option of "RFC2833 enable"..If its there please selct the tick box and reset the phone then test again

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jfernandorosa Tue, 09/03/2013 - 13:21

In ATA187 don´t have this option.

Have (Require DTMF Reception) uncheck.

Thanks!

jfernandorosa Tue, 09/03/2013 - 13:39

Hi,

I attached new log fax_problem_log4.txt.zip

IP ATA187 is 21.10.0.14

In log IP the ATA is ok in RTP.

v=0

o=CiscoSystemsSIP-GW-UserAgent 4424 2824 IN IP4 21.10.0.3

s=SIP Call

c=IN IP4 21.10.0.14  -->IP ATA187

t=0 0

m=audio 16386 RTP/AVP 8 101

c=IN IP4 21.10.0.14

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

Thanks!

Joao

Ayodeji oladipo... Tue, 09/03/2013 - 14:11

You need to look at your ATA confgiuration properly. It looks like its not setup correctly for T38..

When CUBE send a re-INVITE with t38 parameters, CUCM doesnt respond correctly..It send an IP of 0.0.0.0 and no T38 parameterd advertised

Received:

SIP/2.0 200 OK

Via: SIP/2.0/TCP 21.10.0.3:5060;branch=z9hG4bKE3ADC3

From: ;tag=4301F980-385

To: "ATA187 Core" ;tag=24797~fb89236f-816b-47f5-8c94-b8d3c388dd7c-64665313

Date: Tue, 03 Sep 2013 20:30:04 GMT

Call-ID: 96aae900-226146c3-c42-7000a15@21.10.0.7

CSeq: 102 INVITE

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

Allow-Events: presence, kpml

Supported: replaces

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Session-Expires:  1800;refresher=uas

Require:  timer

P-Asserted-Identity: "ATA187 Core"

Remote-Party-ID: "ATA187 Core" ;party=called;screen=yes;privacy=off

Contact: ;+sip.instance="";+u.sip!devicename.ccm.cisco.com="ATA44ADD9D576F4";+u.sip!model.ccm.cisco.com="550"

Content-Type: application/sdp

Content-Length: 240

v=0

o=CiscoSystemsCCM-SIP 24797 2 IN IP4 21.10.0.7

s=SIP Call

c=IN IP4 0.0.0.0

b=TIAS:64000

b=AS:64

t=0 0

m=audio 16386 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000

a=ptime:20

a=inactive

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

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j.huizinga Wed, 09/04/2013 - 04:21

I also had several ATA187 which worked perfectly with an ISDN PRI line.

Then customer switched to SIP trunk and faxes didn't work anymore.

Didn't want to spend to much time on it and configured pass-thru, and faxes work perfectly again.

JH

j.huizinga Wed, 09/04/2013 - 07:26

You have to change to pass-thru on the ATA187 and the cube

fax protocol pass-through

Bye,

Jan

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