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New Member

FoIP - SIP trunk & T.38

We created a SIP trunk from our CUCM to a FAX server.

The FAX server requires T.38 and we created T.38 on a MGCP gateway.

The traffic arrives at the FAX server from the CUCM SIP trunk but nothing is received on the FAX server from the MGCP gateway.

 

How do we bind the SIP trunk to the MGCP gateway to initiate T.38 for the FAX server?

Can the CUCM convert a G711 fax to T.38 without a gateway, if so how?

Attach a .PNG file with the diagram

21 REPLIES
Cisco Employee

FoIP - SIP trunk & T.38

Hi Henry,

     So if I understand correctly; incoming faxes over the MGCP gateway do not reach the fax server on the other end of the SIP Trunk from the CUCM?

Which makes your call flow: PRI >> MGCP Gateway >> CUCM >> SIP >> Fax server.

Also, to answer the last part of your query:

> All fax cals start as Audio calls and then get stepped up to a Fax call.

> The recieving fax machine escalates the call to a fax call; mostly using a method called the Protocol Switchover.

> In our case, for an incoming fax the fax server will send the CUCM a SIP Re-INVITE and will negotiate T.38 with the CUCM.

> The CUCM will negotiate the same using a MGCP MDCX ( Modify Connection ) and if everything works out, the voice call will escalate to a fax call.

> If we had two fax servers on both the ends of the CUCM, it would still be able to escalate the call to a fax call, regardless if agteway is involved.

Please confirm the call flow and I shall then let you know if I need more information on this. Please also include the "show run" from the gateway.

Thank you.

Regards,

Jagpreet

New Member

FoIP - SIP trunk & T.38

Jagpreet

Thank you. I will send the "show run" as soon as possible.

Regards

Henry

New Member

FoIP - SIP trunk & T.38

Jagpreet

The fax flow is as follows.

Incomming FAX:

FAX Machine - PRI - Voice Gateway - MGCP - CUCM - SIP Trunk - FAX Server (T.38)

Outgoing FAX:

FAX server (T.38) - SIP Trunk - CUCM - MGCP - Voice Gateway - PRI - FAX Machine

"SH RUN"

isdn switch-type primary-ntt

!

!

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

sip

!

voice-card 0

dspfarm

dsp services dspfarm

!

voice-card 1

dspfarm

dsp services dspfarm

!

fax interface-type fax-mail

!

controller E1 0/0/1

framing NO-CRC4

pri-group timeslots 1-31 service mgcp

description Telkom FAX PRI Cct number: 0126822000

!

interface GigabitEthernet0/0

ip address 172.18.2.7 255.255.255.0

ip flow ingress

duplex full

speed 1000

h323-gateway voip bind srcaddr 172.18.2.7

!

interface Serial0/0/1:15

description Telkom FAX PRI Cct number: 0126822000

no ip address

encapsulation hdlc

isdn switch-type primary-net5

isdn incoming-voice voice

isdn bind-l3 ccm-manager

no cdp enable

!

voice-port 0/0/1:15

!

ccm-manager fallback-mgcp

ccm-manager redundant-host 172.18.10.23

ccm-manager mgcp

no ccm-manager fax protocol cisco

ccm-manager music-on-hold

ccm-manager config server 172.18.10.21 172.18.10.23

ccm-manager config

!

mgcp

mgcp call-agent 172.18.10.21 2427 service-type mgcp version 0.1

mgcp dtmf-relay voip codec all mode out-of-band

mgcp rtp unreachable timeout 1000 action notify

mgcp modem passthrough voip mode nse

mgcp package-capability rtp-package

mgcp package-capability sst-package

mgcp package-capability pre-package

mgcp default-package fxr-package

no mgcp package-capability res-package

no mgcp timer receive-rtcp

mgcp sdp simple

mgcp fax rate 14400

mgcp fax t38 nsf 000000

mgcp rtp payload-type g726r16 static

mgcp bind control source-interface GigabitEthernet0/0

mgcp bind media source-interface GigabitEthernet0/0

!

mgcp profile default

!

sccp local GigabitEthernet0/0

sccp ccm 172.18.10.21 identifier 1 version 6.0

sccp ccm 172.18.10.23 identifier 2 version 6.0

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 1 register CENT-VGW1-XCOD

!

sccp ccm group 2

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 2 register CENT-VGW1-CFB

!

dspfarm profile 1 transcode

codec g729br8

codec g729r8

codec g711ulaw

codec g711alaw

maximum sessions 30

associate application SCCP

!

dspfarm profile 2 conference

codec g729ar8

codec g729br8

codec g729r8

codec g711ulaw

codec g711alaw

maximum sessions 8

associate application SCCP

!

dial-peer voice 100 voip

destination-pattern 2...

session protocol sipv2

session target ipv4:172.18.22.9

dtmf-relay rtp-nte

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback cisco

!

!

gateway

timer receive-rtp 1200 isdn switch-type primary-ntt
!
!
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
!
voice-card 0
dspfarm
dsp services dspfarm
!
voice-card 1
dspfarm
dsp services dspfarm
!
fax interface-type fax-mail
!
controller E1 0/0/1
framing NO-CRC4
pri-group timeslots 1-31 service mgcp
description Telkom FAX PRI Cct number: 0126822000
!
interface GigabitEthernet0/0
ip address 172.18.2.7 255.255.255.0
ip flow ingress
duplex full
speed 1000
h323-gateway voip bind srcaddr 172.18.2.7
!
interface Serial0/0/1:15
description Telkom FAX PRI Cct number: 0126822000
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
!
voice-port 0/0/1:15
!
ccm-manager fallback-mgcp
ccm-manager redundant-host 172.18.10.23
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 172.18.10.21 172.18.10.23
ccm-manager config
!
mgcp
mgcp call-agent 172.18.10.21 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
mgcp default-package fxr-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax rate 14400
mgcp fax t38 nsf 000000
mgcp rtp payload-type g726r16 static
mgcp bind control source-interface GigabitEthernet0/0
mgcp bind media source-interface GigabitEthernet0/0
!
mgcp profile default
!
sccp local GigabitEthernet0/0
sccp ccm 172.18.10.21 identifier 1 version 6.0
sccp ccm 172.18.10.23 identifier 2 version 6.0
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 1 register CENT-VGW1-XCOD
!
sccp ccm group 2
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 2 register CENT-VGW1-CFB
!
dspfarm profile 1 transcode
codec g729br8
codec g729r8
codec g711ulaw
codec g711alaw
maximum sessions 30
associate application SCCP
!
dspfarm profile 2 conference
codec g729ar8
codec g729br8
codec g729r8
codec g711ulaw
codec g711alaw
maximum sessions 8
associate application SCCP
!
dial-peer voice 100 voip
destination-pattern 2...
session protocol sipv2
session target ipv4:172.18.22.9
dtmf-relay rtp-nte
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback cisco
!
!
gateway
timer receive-rtp 1200

Regards

Henry

New Member

FoIP - SIP trunk & T.38

Jagpreet

Thank you. I will try it tomorrow and let you know.

Thank you for the assistance so far.

Regards

Henry

Cisco Employee

FoIP - SIP trunk & T.38

Hello Henry,

     Any luck with the fax calls after the changes?

Regards,

Jagpreet

New Member

FoIP - SIP trunk & T.38

Hello Jagpreet

No luck yet. Busy collecting the CUCM traces. Will attach them tomorrow morning.

Thank you

Henry

New Member

FoIP - SIP trunk & T.38

Hello Jagpreet

How do we tell the SIP Trunk on the CUCM 9 to use the MGCP gateway?

We do not see any traffic from the gateway to the FAX server.

The CUCM traces.

INVITE sip:1680@172.18.22.9:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.10.21:5060;branch=z9hG4bK1b1126934f7a4
From: <0827870874>;tag=555019~32cbe58c-b967-436d-b4b4-945270831153-36792489
To: <1680>
Date: Wed, 06 Nov 2013 11:12:00 GMT
Call-ID: 40734580-27a12400-127fc-150a12ac@172.18.10.21
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM9.1
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,X-cisco-original-called
Call-Info: <172.18.10.21:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 1081296256-0000065536-0000000424-0352981676
Session-Expires:  1800
P-Asserted-Identity: <0827870874>
Remote-Party-ID: <0827870874>;party=calling;screen=yes;privacy=off
Contact: <0827870874>
Max-Forwards: 70
Content-Length: 0


SIP/2.0 100 Trying
Call-ID: 40734580-27a12400-127fc-150a12ac@172.18.10.21
CSeq: 101 INVITE
From: <0827870874>;tag=555019~32cbe58c-b967-436d-b4b4-945270831153-36792489
To: <1680>
Via: SIP/2.0/UDP 172.18.10.21:5060;received=172.18.10.21;branch=z9hG4bK1b1126934f7a4
Content-Length: 0


SIP/2.0 180 Ringing
Allow: INVITE,BYE,ACK,OPTIONS,CANCEL
Call-ID: 40734580-27a12400-127fc-150a12ac@172.18.10.21
Contact: <1680>
CSeq: 101 INVITE
From: <0827870874>;tag=555019~32cbe58c-b967-436d-b4b4-945270831153-36792489
Server: Netbricks-Sip-T.38IAF/v2.8.0 (Fenestrae, (14 Dec 2010))
To: <1680>;tag=7E3-F9
Via: SIP/2.0/UDP 172.18.10.21:5060;received=172.18.10.21;branch=z9hG4bK1b1126934f7a4
Content-Length: 0


SIP/2.0 200 OK
Allow: INVITE,BYE,ACK,OPTIONS,CANCEL
Call-ID: 40734580-27a12400-127fc-150a12ac@172.18.10.21
Contact: <1680>
Content-Type: application/sdp
CSeq: 101 INVITE
From: <0827870874>;tag=555019~32cbe58c-b967-436d-b4b4-945270831153-36792489
Server: Netbricks-Sip-T.38IAF/v2.8.0 (Fenestrae, (14 Dec 2010))
To: <1680>;tag=7E3-F9
Via: SIP/2.0/UDP 172.18.10.21:5060;received=172.18.10.21;branch=z9hG4bK1b1126934f7a4
Content-Length: 209

v=0
o=Ampath 2890844526 890844526 IN IP4 netbricks.com
s=-
c=IN IP4 172.18.22.9
t=0 0
m=audio 10012 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
m=audio 10012 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20

ACK sip:1680@172.18.22.9:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.10.21:5060;branch=z9hG4bK1b11383ec72d
From: <0827870874>;tag=555019~32cbe58c-b967-436d-b4b4-945270831153-36792489
To: <1680>;tag=7E3-F9
Date: Wed, 06 Nov 2013 11:12:00 GMT
Call-ID: 40734580-27a12400-127fc-150a12ac@172.18.10.21
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0


BYE sip:1680@172.18.22.9:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.10.21:5060;branch=z9hG4bK1b11415b89dd
From: <0827870874>;tag=555019~32cbe58c-b967-436d-b4b4-945270831153-36792489
To: <1680>;tag=7E3-F9
Date: Wed, 06 Nov 2013 11:12:00 GMT
Call-ID: 40734580-27a12400-127fc-150a12ac@172.18.10.21
User-Agent: Cisco-CUCM9.1
Max-Forwards: 70
P-Asserted-Identity: <0827870874>
CSeq: 102 BYE
Reason: Q.850;cause=47
Content-Length: 0


SIP/2.0 200 OK
Allow: INVITE,BYE,ACK,OPTIONS,CANCEL
Call-ID: 40734580-27a12400-127fc-150a12ac@172.18.10.21
Contact: <1680>
CSeq: 102 BYE
From: <0827870874>;tag=555019~32cbe58c-b967-436d-b4b4-945270831153-36792489
Server: Netbricks-Sip-T.38IAF/v2.8.0 (Fenestrae, (14 Dec 2010))
To: <1680>;tag=7E3-F9
Via: SIP/2.0/UDP 172.18.10.21:5060;received=172.18.10.21;branch=z9hG4bK1b11415b89dd
Content-Length: 0

Another trace

INVITE sip:1680@172.18.22.9:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.10.21:5060;branch=z9hG4bK1b11d2f5e47b7
From: "Manie Beneke" <0126781605>;tag=555034~32cbe58c-b967-436d-b4b4-945270831153-36792534
To: <1680>
Date: Wed, 06 Nov 2013 11:12:15 GMT
Call-ID: 49641700-27a1240f-12804-150a12ac@172.18.10.21
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM9.1
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,X-cisco-original-called
Call-Info: <172.18.10.21:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 1231296256-0000065536-0000000425-0352981676
Session-Expires:  1800
P-Asserted-Identity: "Manie Beneke" <0126781605>
Remote-Party-ID: "Manie Beneke" <0126781605>;party=calling;screen=yes;privacy=off
Contact: <0126781605>;video;audio
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 433

v=0
o=CiscoSystemsCCM-SIP 555034 1 IN IP4 172.18.10.21
s=SIP Call
c=IN IP4 172.18.143.167
b=TIAS:768000
b=AS:768
t=0 0
m=audio 25280 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 25320 RTP/AVP 97
b=TIAS:760000
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=420014;packetization-mode=0;max-mbps=36000;max-fs=1280;level-asymmetry-allowed=1
a=content:main

SIP/2.0 100 Trying
Call-ID: 49641700-27a1240f-12804-150a12ac@172.18.10.21
CSeq: 101 INVITE
From: "Manie Beneke" <0126781605>;tag=555034~32cbe58c-b967-436d-b4b4-945270831153-36792534
To: <1680>
Via: SIP/2.0/UDP 172.18.10.21:5060;received=172.18.10.21;branch=z9hG4bK1b11d2f5e47b7
Content-Length: 0


SIP/2.0 180 Ringing
Allow: INVITE,BYE,ACK,OPTIONS,CANCEL
Call-ID: 49641700-27a1240f-12804-150a12ac@172.18.10.21
Contact: <1680>
CSeq: 101 INVITE
From: "Manie Beneke" <0126781605>;tag=555034~32cbe58c-b967-436d-b4b4-945270831153-36792534
Server: Netbricks-Sip-T.38IAF/v2.8.0 (Fenestrae, (14 Dec 2010))
To: <1680>;tag=7E3-E224
Via: SIP/2.0/UDP 172.18.10.21:5060;received=172.18.10.21;branch=z9hG4bK1b11d2f5e47b7
Content-Length: 0


SIP/2.0 200 OK
Allow: INVITE,BYE,ACK,OPTIONS,CANCEL
Call-ID: 49641700-27a1240f-12804-150a12ac@172.18.10.21
Contact: <1680>
Content-Type: application/sdp
CSeq: 101 INVITE
From: "Manie Beneke" <0126781605>;tag=555034~32cbe58c-b967-436d-b4b4-945270831153-36792534
Server: Netbricks-Sip-T.38IAF/v2.8.0 (Fenestrae, (14 Dec 2010))
To: <1680>;tag=7E3-E224
Via: SIP/2.0/UDP 172.18.10.21:5060;received=172.18.10.21;branch=z9hG4bK1b11d2f5e47b7
Content-Length: 244

v=0
o=Ampath 2890844526 890844527 IN IP4 netbricks.com
s=-
c=IN IP4 172.18.22.9
t=0 0
m=audio 0 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
m=video 10012 RTP/AVP 97
b=TIAS:39104

ACK sip:1680@172.18.22.9:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.10.21:5060;branch=z9hG4bK1b11e4e8c898d
From: "Manie Beneke" <0126781605>;tag=555034~32cbe58c-b967-436d-b4b4-945270831153-36792534
To: <1680>;tag=7E3-E224
Date: Wed, 06 Nov 2013 11:12:15 GMT
Call-ID: 49641700-27a1240f-12804-150a12ac@172.18.10.21
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0


BYE sip:0126781605@172.18.10.21:5060 SIP/2.0
Allow: INVITE,BYE,ACK,OPTIONS,CANCEL
Call-ID: 49641700-27a1240f-12804-150a12ac@172.18.10.21
CSeq: 2 BYE
From: <1680>;tag=7E3-E224
Max-Forwards: 70
To: "Manie Beneke" <0126781605>;tag=555034~32cbe58c-b967-436d-b4b4-945270831153-36792534
User-Agent: Netbricks-Sip-T.38IAF/v2.8.0 (Fenestrae, (14 Dec 2010))
Via: SIP/2.0/UDP 172.18.22.9:5060;branch=z9hG4bK51A5F06
Content-Length: 0


SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.22.9:5060;branch=z9hG4bK51A5F06
From: <1680>;tag=7E3-E224
To: "Manie Beneke" <0126781605>;tag=555034~32cbe58c-b967-436d-b4b4-945270831153-36792534
Date: Wed, 06 Nov 2013 11:12:16 GMT
Call-ID: 49641700-27a1240f-12804-150a12ac@172.18.10.21
CSeq: 2 BYE
Content-Length: 0


Regards

Henry

FoIP - SIP trunk & T.38

Hi,

Do you have the possibility to run Wireshark on your fax server ? If yes look at the SIP SDP message sent from CUCM. You normally should see the IP address of your MGCP gateway as the originating point for the media transfer.

If really you do not see any traffic and it's not a negociation issue then it could be a IP routing issue between your gw and the fax server (no route to destination).

JC

New Member

FoIP - SIP trunk & T.38

Hello JC

Thank you for your reply.

Please see my reply to Jagpreet

Regards

Henry

FoIP - SIP trunk & T.38

Reason: Q.850;cause=47  means codec issue.

Do you have transcoding ressources configured ?

New Member

FoIP - SIP trunk & T.38

It is unclear what you are getting at here. First, you need to be clear if the provider supports t.38 or not. If they don't support T.38, your best bet is to nail up a g.711 call to the provider, and you'll get best-effort faxing which should be fine at low speeds. High speed fax and modems would be sketchy at best across a g.711 call in voice mode. You can use specific device pools/regions/dial-peers to acheive this such that calls from/to the fax numbers always are g711 from the start of the call.

Keep in mind that if you are doing any switchovers via NSE (gw-controlled MGCP, SCCP, or nse-forced) then make sure you aren't invoking the MTP, as MTPs don't pass NSEs properly out the other side. Modem passthrough is Cisco proprietary, so you'll never get that to work with a non-Cisco equipped SP. Furthermore, modem passthrough requires an NSE based switchover, which isn't going to work with the MTP as noted above.

You can invoke an MTP for a T.38 call, as long as 'codec pass-thru' is specified. Obviously, your provider needs to support T.38, though. You need to ensure that you use protocol-based switchovers for this (CA-controlled with MGCP, or via H.323 RM/SIP Re-INVITE messaging.)

Fax pass-through switchover is essentially just an upspeed to g711ulaw once the call is established via the call-control protocol messaging, so you can do this with an MTP.

Whether you are doing SIP or H.323, capabilities are the same on the MTP from a functional perspective, since the MTP is really only speaking SCCP anyway.

As far as DTMF-relay goes, the only thing relevant to talk about is RTP-NTE and in-band audio. All other types of DTMF-relay follow the signaling path, and are irrelevant from the perspective of an MTP (which terminates media). If you need to convert DTMF to-from in-band audio, you need a transcoder. Converting RTP-NTE to another type of DTMF-relay (H245-alpha/signal, SIP notify, etc.) can be done with a software or hardware MTP, as well as a transcoder.

FoIP - SIP trunk & T.38

Your CUCM says BYE to your fax server. You haven't even reached the first step negociation with G711 A-law or mu-law

The other trace involving Manie Beneke shows you use G729

You need to allow G711 codec with your fax server.

You've put your SIP Trunk into a Device Pool. According to this Device pool - specific if well done to your Fax Server - check the associated Region Settings. The "Max Audio Bit Rate" of this region must be set to G.711.

JC

New Member

FoIP - SIP trunk & T.38

Thank you JC. Will check and let you know.

Regards

Henry

New Member

FoIP - SIP trunk & T.38

JC

We made the changes as you mentioned, but still no luck. Below the new trace.

INVITE sip:1680@172.18.22.9:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.10.21:5060;branch=z9hG4bK1b94748acfd76
From: <0827870874>;tag=563017~32cbe58c-b967-436d-b4b4-945270831153-36920033
To: <1680>
Date: Thu, 07 Nov 2013 10:25:08 GMT
Call-ID: dec7df80-27b16a84-12dee-150a12ac@172.18.10.21
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM9.1
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,X-cisco-original-called
Call-Info: <172.18.10.21:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 3737640832-0000065536-0000000437-0352981676
Session-Expires:  1800
P-Asserted-Identity: <0827870874>
Remote-Party-ID: <0827870874>;party=calling;screen=yes;privacy=off
Contact: <0827870874>
Max-Forwards: 70
Content-Length: 0


SIP/2.0 100 Trying
Call-ID: dec7df80-27b16a84-12dee-150a12ac@172.18.10.21
CSeq: 101 INVITE
From: <0827870874>;tag=563017~32cbe58c-b967-436d-b4b4-945270831153-36920033
To: <1680>
Via: SIP/2.0/UDP 172.18.10.21:5060;received=172.18.10.21;branch=z9hG4bK1b94748acfd76
Content-Length: 0


SIP/2.0 180 Ringing
Allow: INVITE,BYE,ACK,OPTIONS,CANCEL
Call-ID: dec7df80-27b16a84-12dee-150a12ac@172.18.10.21
Contact: <1680>
CSeq: 101 INVITE
From: <0827870874>;tag=563017~32cbe58c-b967-436d-b4b4-945270831153-36920033
Server: Netbricks-Sip-T.38IAF/v2.8.0 (Fenestrae, (14 Dec 2010))
To: <1680>;tag=7E3-82D6
Via: SIP/2.0/UDP 172.18.10.21:5060;received=172.18.10.21;branch=z9hG4bK1b94748acfd76
Content-Length: 0


SIP/2.0 200 OK
Allow: INVITE,BYE,ACK,OPTIONS,CANCEL
Call-ID: dec7df80-27b16a84-12dee-150a12ac@172.18.10.21
Contact: <1680>
Content-Type: application/sdp
CSeq: 101 INVITE
From: <0827870874>;tag=563017~32cbe58c-b967-436d-b4b4-945270831153-36920033
Server: Netbricks-Sip-T.38IAF/v2.8.0 (Fenestrae, (14 Dec 2010))
To: <1680>;tag=7E3-82D6
Via: SIP/2.0/UDP 172.18.10.21:5060;received=172.18.10.21;branch=z9hG4bK1b94748acfd76
Content-Length: 209

v=0
o=Ampath 2890844526 890844526 IN IP4 netbricks.com
s=-
c=IN IP4 172.18.22.9
t=0 0
m=audio 10012 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
m=audio 10012 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20

ACK sip:1680@172.18.22.9:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.10.21:5060;branch=z9hG4bK1b9487405e87c
From: <0827870874>;tag=563017~32cbe58c-b967-436d-b4b4-945270831153-36920033
To: <1680>;tag=7E3-82D6
Date: Thu, 07 Nov 2013 10:25:08 GMT
Call-ID: dec7df80-27b16a84-12dee-150a12ac@172.18.10.21
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0


BYE sip:1680@172.18.22.9:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.10.21:5060;branch=z9hG4bK1b94937a13016
From: <0827870874>;tag=563017~32cbe58c-b967-436d-b4b4-945270831153-36920033
To: <1680>;tag=7E3-82D6
Date: Thu, 07 Nov 2013 10:25:08 GMT
Call-ID: dec7df80-27b16a84-12dee-150a12ac@172.18.10.21
User-Agent: Cisco-CUCM9.1
Max-Forwards: 70
P-Asserted-Identity: <0827870874>
CSeq: 102 BYE
Reason: Q.850;cause=47
Content-Length: 0


SIP/2.0 200 OK
Allow: INVITE,BYE,ACK,OPTIONS,CANCEL
Call-ID: dec7df80-27b16a84-12dee-150a12ac@172.18.10.21
Contact: <1680>
CSeq: 102 BYE
From: <0827870874>;tag=563017~32cbe58c-b967-436d-b4b4-945270831153-36920033
Server: Netbricks-Sip-T.38IAF/v2.8.0 (Fenestrae, (14 Dec 2010))
To: <1680>;tag=7E3-82D6
Via: SIP/2.0/UDP 172.18.10.21:5060;received=172.18.10.21;branch=z9hG4bK1b94937a13016
Content-Length: 0


Regards

Henry

VIP Super Bronze

FoIP - SIP trunk & T.38

Henry,

What is the region setting between the gateway and the sip trunk set to? From the trace it looks like it is set to G729. Your fax server wants to do either G711u or G711alaw..But CUCM cant do any of that codec, this is most likely due to region settings..Please check this and ensure that it is set to G711

Please rate all useful posts

"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

FoIP - SIP trunk & T.38

The region is set to G711. If you look at the latest attach trace "0 PCMU/8000 a 0 PCMU/8000 a" means that we use G711. Where in the trace do you still see G729.

Thank you

Regards

Henry

VIP Super Bronze

FoIP - SIP trunk & T.38

That is what your fax server is advertising. That is the 200 OK from your fax server. CUCM needs to send an ACK to that 200 ok and the codec used will depend on the region between the gateway and the sip trunk to the fax server. Can you send the full cucm traces?

Please rate all useful posts

"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

FoIP - SIP trunk & T.38

I will ask the customer for the traces and will forward them to you ASAP. That will most likely be later tonight or tomorrow.

Thank you

Regards

Henry

FoIP - SIP trunk & T.38

"0 PCMU/8000 a 0 PCMU/8000 a" is what your fax server asks to CUCM, which is unable to do so because the Region is not allowed to use G711.

If you have configured this setting into the Region have you reset the SIP Trunk to apply changes ?

New Member

FoIP - SIP trunk & T.38

We will setup a test environment tomorrow. We will configure it separately from the life environment and then run the test. Will update as soon as the tests are finish.

Thank you all for the assistance so far.

Regards

Henry

Cisco Employee

Re:FoIP - SIP trunk & T.38

Henry,

As Aok said, we need the complete CCM traces to move forward on this.

Regards,

Jagpreet


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