Re: Ask the Expert: Configuring and Troubleshooting Faxing in Ci

ciscomoderator · ‎11-02-2012

With Edson Pineiro

Welcome to the Cisco Support Community Ask the Expert conversation. This is an opportunity to learn best practices and tech tips from Cisco expert Edson Pineiro on the common and complex issues with faxing in Cisco IP voice networks. Cisco fax networks run on top of signalling protocols such as fax T.38, T.37, pass-through, NSE based modem passthrough and the underlying fax T.30 protocol with different modulation types. You can ask any questions on how Cisco Fax networks interact with any of the signalling protocols mentioned before.

Edson Pineiro is a senior customer support engineer in the Cisco Technical Assistance Center in Sydney. His current role includes configuring, troubleshooting, and designing gateways, gatekeepers, Cisco Unified Border Element Enterprise Edition, and Cisco Unified Call Manager using his deep knowledge of signaling protocols such as SIP, H.323, MGCP, SKINNY, and others. He has been involved in several bug fixes, escalations, and critical account cases from around the globe. He has over seven years of experience in the IP voice industry.

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Edson might not be able to answer each question due to the volume expected during this event. Remember that you can continue the conversation on the Collaboration, Voice and Video sub-community forum shortly after the event. This event lasts through Nov 16, 2012. Visit this forum often to view responses to your questions and the questions of other community members.

Mizanul Islam · ‎11-05-2012

Hi Edson Pineiro,

We got a new project of University campus area network where Products has (Cisco voice gateway router

C2921-CME-SRST/K9 ,

FL-CME ,FL-CME-SRST- 50, with

VWICC2-1MFT-T1/E1 , and

CP-7911G IP phone one Internet router

C2911-Sec/K9) all access switch are POE.

Can you share me a simple configuration example for deploy this network. No redundant devices.

Regards

Rasel

Edson Pineiro · ‎11-05-2012

Hello Rasel,

Thank you for your question. Unfortunately the scope of this forum is with regards to faxing and your device list does not include a fax machine.

I may have to refer you to the Call Manager Express Administration Guide:

Cisco Unified CME Overview:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeover.html

Please also post your specific Call Manager Express related questions in the following forum:

Cisco IP Telephony Forum:

https://supportforums.cisco.com/community/netpro/collaboration-voice-video/ip-telephony

To answer your question please find the following sample Telephony-Service Configuration Sample:

!

telephony-service

auto assign 1 to 5

max-ephones 22

max-dn 88

ip source-address port 2000

max-redirect 20

time-zone 47

max-conferences 8 gain -6

call-forward pattern .

transfer-system full-consult

create cnf-files

!

ephone-dn 1 dual-line

number 1001

!

ephone-dn 2 dual-line

number 1002

!

ephone-dn 3 dual-line

number 1003

!

ephone-dn 4 dual-line

number 1004

!

ephone-dn 5 dual-line

number 1005

!

I am assuming your using Call Manager Express since you did not list an MCS server. Also you need to at least configure some IP addressing on the CME router interface and add it into the above configuration sample. You can refer to the CME administration guide for a configuration command reference to setup DHCP with option 150. You can also configure a static IP on the phone for testing don't forget to point the tftp server (options 150) IP address to the telephony-services source IP address configured above.

This is as much as I can assist you without moving away from the topic of the forum. If you have any faxing related questions I can continue to assist.

Sound like a fun project, good luck :-)

Thank you

&

Regards

Edson Pineiro

CISCO

Mohamed Wahba · ‎12-18-2013

Hello Edson,

I have a recently problem with fax has came to surface
All fax outgoing are working fine
All internal faxes are working fine
Most of incoming faxes from pstn are fail, and may be after more than one trial work. After debugging on voice gateways i noticed that the gateway sends disconnect isdn message to the pstn incoming fax.

Sent from Cisco Technical Support iPad App

Novriadi . · ‎11-05-2012

Hi Edson

we will deploy fax ip (Xerox) on IP Voice environment (Call Manager), we have test Xerox IP Fax (DocuCentre X Series) to directly registered to Call Manager using SIP (3rd party basic).

I attached the scheme to make you clear, and several configuration of my Router Voice Gateway.

When we test :

CM version : 8.6

1. Fax call between Xerox IP FAX : OK (internal call).

2. Fax call to PSTN : OK (outgoing call).

3. Fax call from PSTN to Xerox IP Fax : Not OK (incoming call).

but if we try to call to SIP Phone from PSTN the result is OK.

from debug capture we get the information that the fax is always busy if we call from PSTN.

we also test with CM version 6 with same configuration, all of test status are OK.

Is there any misconfiguration or compatibility issue ?

thanks for your help.

Edson Pineiro · ‎11-06-2012

Hello Novriadi,

Thank you for providing logs. I reviewed the debugs of both the test phone call and the Fax call to the extension 6000. When the call terminates on the fax machine the T38 protocol does not negotiate. According to the debugs on the gateway, the gateway receives a 200-OK without the t38 media included in the SDP. The gateway responds to the 200-OK with an ACK including the t38 media in its SDP, however the ACK response from the call manager does not include the t38 media in its SDP. Please see the following snippets:

!

Received:

SIP/2.0 200 OK

a=rtpmap:0 PCMU/8000

!### ------ There is no T38 SDP media included in this 200-OK.

!

Sent:

ACK sip:6000@172.16.1.200:5060 SIP/2.0

m=image 16406 udptl t38

!

Received:

ACK sip:172.16.1.1:5060 SIP/2.0

!### ------ There is no T38 SDP media included in this ACK.

!

It would be interesting to see if the ACK+SDP transmitted from the gateway to the call manager is forwarded to the IP fax machine. If the CUCM is dropping the t38 SDP media attribute towards the IP fax machine then the fax machine may consider the call to be passthrough using g711ulaw. In this scenario protocol based passthrough is not configured on the gateway.

Regardless of the above signalling, the basic function of any fax protocol is to establish a clear voice call before up-speeding. This means that the fax call will negotiate basic audio call, before negotiating the fax protocols. Once the voice call is established between the Originating gateway (in this case the SIP gateway) and the terminating IP fax endpoint (TGW) Terminating gateway, the terminating fax endpoint should initiate a RE-INVITE including the SDP t38 media attributes. The term used for this fax media re-negotiation is called fax codec up-speeding and it's all ways up to the terminating fax device to initiate the up-speed RE-INVITE. The SIP signalling flow may look like so:

!

!!###Establish a call with a voice codec first

!

Fax -----> OGW --------------------------- SIP INVITE -------------------------->TGW -------------------> Fax

Fax-------- OGW ------------------------- SIP 180 Ring <--------------------------TGW -------------------- Fax

Fax ------- OGW -------------------- SIP 200OK +SDP Ulaw <-----------------TGW ----------------- > Fax

Fax ------- OGW -------------------- SIP ACK + SDP Ulaw -------------------->TGW ---------------- > Fax

Fax <--------------------------- Basic Voice Call with RTP established -------------------------------> Fax

Fax -------------------> Originating fax machine send a T30 CNG tone ------------------> Fax

Fax <---- Terminating fax machine detects a T30 CNG, sends a CED tone, initiates a RE-INVITE & upspeed <--- Fax

!

!!### Terminating IP fax endpoint up-speeds the voice call with a T.38 codec after detecting the CNG tone and sending a CED tone.

!

Fax --------- OGW ----------------------- SIP INVITE + t38 SDP <-------- TGW ---------------------- Fax

Fax -------> OGW --------------------- SIP 180 Ring -------------------------->TGW -------------------- > Fax

Fax -------> OGW --------------------- SIP 200 OK + t38 SDP ------------ TGW --------------------- > Fax

Fax -------> OGW <--------------------- SIP ACK <------------------------------TGW --------------------- > Fax

Fax<->Now that the T38 protocol has been negotiated we can start the T30 signalling Transmission<->Fax

!

!!###Establish, negotiate a T30 fax capabilities and transmit fax data.

!

Fax ---> OGW ------------------------------ CNG (Calling Tone) 1100 hz ------------------------->TGW ------ > Fax

Fax ---- OGW <----------------- CED (Called Terminal Identification) 2100 hz <-------------TGW <------ Fax

Fax ---- OGW <--------- DIS (Digital Identification Signal) + Optional NSF &CSI <----- TGW <------ Fax

Fax --> OGW ---------------- DCS (Digital Command signal) + Optional TSI ------------->TGW ------ > Fax

Fax --> OGW -------------------------------- TCF (training Check) ---------------------------------->TGW ------ > Fax

Fax --- OGW <------------------------- CFR (confirmation to Receive) <-------------------------TGW <------- Fax

Fax -> OGW ----------------- PPT/PPS-NULL (partial page transmission) ----------------->TGW ------- > Fax

Fax --- OGW <------------------------- MCF (Message Confirmation) <---------------------------TGW <------- Fax

Fax -> OGW -------------------------- PPS-MPS (partial page sent) ---------------------------->TGW ------- > Fax

Fax --- OGW <----------------------- MCF (Message Confirmation) <----------------------------TGW <------- Fax

Fax -> OGW -------------- PPS-EOP (Partial page sent) (end of page) -------------------->TGW ------- > Fax

Fax --- OGW <----------------------- MCF (Message Confirmation) <-----------------------------TGW <------- Fax

Fax -> OGW ----------------------------------- DCN (disconnect) ------------------------------------>TGW ------- > Fax

Given the above sample voice and fax signalling call flows, the missing part to the puzzle is the fax codec up-speed. We can successfully establish a voice call with the IP fax machine however we do not see the IP fax initiating an up-speed with a RE-INVITE +T.38 SDP media attribute. The only way to verify why the IP Fax machine does not up-speed is as though the originating fax machine does not send a CNG tone or the terminating fax does not receive the CNG tone and if it does and it is still not initiating a RE-INVITE +SDP, ask yourself does my terminating IP fax machine support T.38 or is it using protocol based fax passthrough?

The only way to confirm if the IP fax machine received the CNG T30 fax tone is to capture a packet capture on the terminating fax machine, decode, analyse and export the RTP traffic to a audio format then listen to the audio for the CNG tone. When analysing the CNG tone, the frequency spectrum should show that the tone frequency is 1100hz.

Also the packet capture on the IP fax machine will verify if this fax machine is sending a busy tone before or after receiving the CNG tone.

With regards to your IOS configuration, third party devices do not support Cisco Fax relay. In fact Cisco Fax relay has stability issues hence it has been removed starting from IOS 15.x onwards. May I suggest the following tests and changes on the SIP gateway:

1. For testing please try protocol based passthrough:

!

voice service voip

no modem passthrough nse codec g711ulaw

fax protocol pass-through g711ulaw

!

dial-peer voice x voip

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

fax protocol system

fax rate 14400

fax-relay ecm disable

!

The reason why I would like to disable nse based modem passthrough because it is a cisco proprietary fax and modem protocol and will not be supported by third party devices. I'm also disabling error correction mode because it causes too many fax failures during transmission and the retransmissions time out, causing intermittent failures on the fax machine when there is minor QOS issues on the rtp stream. The fax rate 14400 is recommended because the default Cisco IOS behaviour enables SG3 spoofing by blocking the CM tone during high speed super group 3 speed negotiation.

2. After verifying if protocol based passthrough is not working, test the following t.38 settings:

!

voice service voip

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

!

dial-peer voice x voip

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

fax protocol system

fax rate 14400

fax-relay ecm disable

!

1. If the above does not work I would like to compare the following debugs for a working outbound fax and non working incoming fax using t.38 and protocol based passthrough.

!

deb fax relay t30 all

debug ccsip messages

debug voip ccapi

!

2. Please also upload the packet capture for both the working and failed scenario from the IP fax machine.

In summary, due to so many different capabilities enabled on the end fax endpoints (physical fax machine), standards, pre-standards, drafts, proprietary vendor based standards, and different signalling methods to relay and switch to the fax codec, there is a little confusion on how it will all integrate with VOIP. Faxing has been around since the first Facsimile Apparatus by Alexander Bain back in 1893 using a pendulum to scan dark spaces on a page and send electrical pulse. Faxing has come along way since then and I'm still looking forward to the new 3d printing manufacturers to include a fax line on there printers.

Please let me know if you have any questions.

Thank you

&

Regards

Edson Pineiro

CISCO

Daniele Giordano · ‎11-06-2012

Hi Edson, I'm working with AS5400s 12.4.15T15 controlled via MGCP by PGW.

In our scenario we use fax pass-through triggered by NSE events.

Fax transmissions are generally OK but there is a little percentage of failed fax.

Tipically faxes are handled via Cisco NSE 192 events. We studied many traces and we observed that failed faxes occur when NSE 192 event is followed by a NSE 193 event.

Our fax speed is 9600 or 14400 bps (V.34 and ECM off), so the NSE 193 event that handles echo cancellation it doesn't required.

NSE 193 should be triggered by ANSam tone, similar to 2100 Hz CED tone, but with phase reversal every 450 ms.

Is it possible that AS5400 confuses 2100 Hz tone?

Is there configuration trick?

Is there a voicecap Vregister to change?

Best Regards.

Edson Pineiro · ‎11-07-2012

Hi Daniele,

Thank you for your question. When the NSE 193 follows a NSE 192, the destination fax machine signals the calling fax with an Ansam (2100hz) + phase reversal. The NSE 192 is triggered by an Ansam which disables silence suppression on the call-leg however the NSE payload 193 is triggered by the Ansam including a phase reversal. The NSE payload 193 disables the echo-cancellers on the call-leg which is needed to pass-through fax modulation types such as V.34 for super G3 speeds. The regular Ansam excluding the phase reversal is used to train modulation types V.32 or V.23 etc, Group 3 Speed fax machines hence we only disable silence suppression when signalling the peer with an NSE 192.

Your are correct by saying that the NSE 193 should be triggered by an Ansam (2100 hz) tone with phase reversal every 450ms. However why are your fax machines trying to use super group 3 (SG3) speeds of up to 33.6 kbs when your fax speed is only set to 9600 or 14400 bps?

To verify if in fact the AS5400 is confused, you need to confirm if the Ansam received from the terminating fax machine is sending the Ansam with phase reversal and if it is, the router is functioning as designed, because NSE based modem passthrough supports SG3. I wonder what the issue is when using SG3 speeds...

Depending on the type of dsp used on the AS5400 will depend if you can capture a PCM trace. The PCM capture will allow me to analyse the raw audio sent through the dsp. I should be able to check the Ansam frequency from the audio file using a spectrum analyser. Otherwise a sniffer capture can also help analyse the embedded audio from the RTP packet. Regardless, the only way the IOS will send an NSE 193 is as if it was triggered with an SG3 Ansam including phase reversal. If your sure that you are using 9600 and 14400 bps then you should verify the fax or other voip settings to confirm if this is the case.

If you have any further questions please let me know.

Thank you

&

Regards

Edson Pineiro

CISCO

Daniele Giordano · ‎11-07-2012

Hi Edson, thank you very much for the precious answer. Can you give me a suggestion about a modem problem?

I've this working scenario:

Cisco ATA SPA2102 ---> SIP Server ---> Cisco 1760 PSTN gateway connected to an ISDN BRI

The modem pass through is triggered by NSE and all works fine.

I would use a more robust and flexible PSTN gateway: PGW with a stack of cisco AS5400s connected to an ISUP trunk.

Cisco SPA2102 ---> SIP Server ---> PGW (signalling) /AS5400 (media) connected to ISUP trunk

In this case the same call fails.

I spent much time to find the cause but I wasn't lucky.

The difference in traces is in the call closing phase. Can be an issue?

Best Regards.

Edson Pineiro · ‎11-08-2012

Hi Daniele,

Thank you for providing and collecting captures.

I analysed both the packet captures and found that in both the working and failed scenarios the modem call did not successfully negotiate and exchange NSE based modem passthrough. According to the capture I could only see the NSE event 192 being initiated by IP address 10.28.32.190, however neither the as5400 or the 1760 responded to the 192 request with a 192 response. You can use the following filter in wireshark "rtpevent.event_id == 192" to check the 192 exchange or "rtpevent.event_id == 192 && rtpevent.event_id == 193" to check both the 192 and 193 exchange.

Please see the following print screen with regards to my reference in the capture:

Also According to the spa2102 data-sheet, there is no mention of modem support on the spa2102, however it does say that the ata supports fax passthrough and t.38.

Cisco SPA2102 Phone Adapter with Router

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps10024/ps10026/data_sheet_c78-502137.html

I ran through a search in the database and fond the following possible issues. However based on the capture I don't believe these are relevant.

CSCtj59199 - srp5xx:fax with PSTN using NSE method,sometimes fail to receive the file

CSCuc79489 - request to ignore NSE event 193 which affects faxing

CSCub93897 - Poor modem performance

It may be that in certain scenarios the spa module successfully sends modems calls, so I decoded the rtp audio using a spectrum analyser. As per the frequency analyser the working modem call is complete from start to end including disconnect tone. However when the modem call terminates on the as5400 interestingly there was no disconnect tone, so the call was incomplete. In order to send a modem call successfully NSE based modem passthrough is needed to exchange 192 and 193 to disable both vad and the ECAN (echo cancellers). Reason being is that when these features are disabled the audio using g711u/alaw is not tampered with by these feature and it's good enough to passthrough the modem tones, since modem and high speed faxing is very sensitive to audio quality it works.

From what I can see, both the working and failed modem call the call is using the best effort in-band modem or in other words what we consider to be in-band faxing. In-band modem or in-band fax basically happens when the voice audio codec is negotiated with a g711 codec and either the modem or the fax protocol signalling fails to negotiate however the voice call remains connected. This is a best effort scenario since neither silence suppression or the echo canceller is disabled. However for the modem or fax call to complete the voice only call needs to remain connected until the modem or fax completes it's transmission.

According to the failed pack capture all 3 test modem calls were disconnected by the far end SIP server before the modem could have finished sending the transmission. However when comparing the c1760, the 3 working test calls were disconnected by the SPA2102 and as per the modem tone decode it was enough time for the disconnect tone to complete the call and transmit the data. Please see the followng print screens taken from both the capture and the audio frequency analysis:

Working audio analysis

It's a complete call you can see the disconnect tone.

Failed audio analysis

This call is incomplete, the audio just drop, when the SIP server sent a bye request.

Working call, the BYE request was sent from the spa2102:

Failed call, the BYE request was sourced from the far end SIP device:

To troubleshoot in-band fax or in-band modem, you need to investigate the voice signalling to confirm why the call is disconnecting and not staying connected long enough to transmit the modem data. However be advised this is a best effort scenario and it's recommended to have a device that supports modem.

To further investigate please collect the following debugs on the as5400:

!

debug voip rtp session named-events

debug mgcp packet

debug voip ccapi

!

Please also collect the following:

!

show mgcp

!

In addition to the above output from the as5400, please provide the spa2102 configuration.

For your reference I'm attaching the audio files taken from the captures.

If you have any further questions please do not hesitate to contact me.

Thank you

&

Regards,

Edson Pineiro

CISCO

bsoth · ‎11-06-2012

I have a 2811 that I want to put all my faxing onto. We have a dedicated PRi for FAX that will be connected to this router and I want to create the Analog (FXS) extensions off this router. I have the router which already has an HDV module with the 1MFT-T1-DI in it. It also have 16 DSP channels. I did the sh voice dsp voice command and found the following:

brs-Voice.GW_1#sh voice dsp voice

DSP DSP DSPWARE CURR BOOT PAK TX/RX

TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABORT PACK CO UNT

==== === == ======== ========== ===== ======= === == ========= == ===== ======== ====

C549 013 01 g729r8 23.8.1 busy idle 0 0 1/0/0:0 19 0 5839782/ 5857

622

02 {medium} 23.8.1 IDLE idle 0 1/0/0:0 20 0 5735478/ 5943

595

03 {medium} 23.8.1 IDLE idle 0 1/0/0:0 21 0 6019741/ 6104

480

04 {medium} 23.8.1 IDLE idle 0 1/0/0:0 22 0 5627646/ 5776

613

C549 014 01 {medium} 23.8.1 IDLE idle 0 0 1/0/0:0 23 0 5174328/ 5308

818

02 {medium} 23.8.1 IDLE idle 0 1/0/0:0 24 0 6065215/ 5867

603

edsp 0001 01 g729r8 p 0.1 IDLE 50/0/1.1

----------------------------FLEX VOICE CARD 0 ------------------------------

*DSP VOICE CHANNELS*

CURR STATE : (busy)inuse (b-out)busy out (bpend)busyout pending

LEGEND : (bad)bad (shut)shutdown (dpend)download pending

DSP DSP DSPWARE CURR BOOT PAK TX/RX

TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK CO UNT

===== === == ========= ========== ===== ======= === == ========= == ==== ======= =====

C5510 001 01 None 23.8.1 idle idle 0 0 0 0/0

C5510 001 02 None 23.8.1 idle idle 0 0 0 0/0

C5510 001 03 None 23.8.1 idle idle 0 0 0 0/0

C5510 001 04 None 23.8.1 idle idle 0 0 0 0/0

C5510 001 05 None 23.8.1 idle idle 0 0 0 0/0

C5510 001 06 None 23.8.1 idle idle 0 0 0 0/0

C5510 001 07 None 23.8.1 idle idle 0 0 0 0/0

C5510 001 08 None 23.8.1 idle idle 0 0 0 0/0

C5510 001 09 None 23.8.1 idle idle 0 0 0 0/0

C5510 001 10 None 23.8.1 idle idle 0 0 0 0/0

C5510 001 11 None 23.8.1 idle idle 0 0 0 0/0

C5510 001 12 None 23.8.1 idle idle 0 0 0 0/0

C5510 001 13 None 23.8.1 idle idle 0 0 0 0/0

C5510 001 14 None 23.8.1 idle idle 0 0 0 0/0

C5510 001 15 None 23.8.1 idle idle 0 0 0 0/0

C5510 001 16 None 23.8.1 idle idle 0 0 0 0/0

------------------------END OF FLEX VOICE CARD 0 ----------------------------

So I will buy the FXS WICs: 4 VIC3-4FXS/DID so this will give me the 16 analog ports I need.

Does anyone see a problem with this config?

jackylee · ‎11-06-2012

Not sure has anyone aware of this document in regads to different fax scenario that has been posted the cummunity:

https://supportforums.cisco.com/docs/DOC-14593

Edson Pineiro · ‎11-07-2012

Hello Bsoth,

With regards to your question will the VIC3-4FXS/DID give you 16 analogue ports. The answer is the 4 port FXS card will only allow a maximum of 4 ports.

If your going to install a T1 module, you should keep in mind that a 24 B channel PRI will consume 24 dsp channels, and if there is only 16 dsp channels there won't be enough channels for both or the PRI or a T1-CAS and or the FXS card.

However if your going to use the T1 for channelised data, then a 16 dsp channel PVDM will be sufficient for a 4 port FXS card but you will only have 4 ports. Please ensure you check if both the PVDM and the FXS card are compatible with the Chassis.

If you are using the T1 for a PSTN connection may I suggest to review the following dsp calculator, to verify how many dsp channels you need for a PSTN T1 and an FXS card.

DSP Calculator:

http://www.cisco.com/web/applicat/dsprecal/dsp_calc.html

Please let me know if you have any further questions.

Thank you

&

Regards,

Edson Pineiro

CISCO

Rob Huffman · ‎11-07-2012

Hi Edson,

WOW! Some great, great answers here! +5 for this soon to be bookmarked event

Cheers!

Rob

"And if I should fall behind
Wait for me" - Springsteen

Roger Kallberg · ‎11-08-2012

I can only agree with what Rob wrote. Really great work Edson (+5) all day long

Please remember to rate helpful responses and identify helpful or correct answers.

Ask the Expert: Configuring and Troubleshooting Faxing in Cisco IP Voice Networks