About weavermj1982

weavermj1982 · ‎07-18-2012

Hello fellow support forum members, Last week I posted an issue I was having where some Verifone chip and pin terminals were failing to negotiate at higher modem speeds. To resolve this, I enabled modem passthrough on our PRI gateway and configured a dial-peer on the VG224 to use modem passthrough and send the call direct to the PRI gateway over IP encoded in G.711ulaw, rather than via the SCCP route. (https://supportforums.cisco.com/message/3683083#3683083) Since applying this config, ATA adapters are now unable to send faxes to the PSTN. I am running ATA186s as follows: S/W Version 3.02.03(051201A) H/W Version 0x0013 0x0000 Product ID ATA186I2-A with Audio mode and Connect mode set on factory defaults as: Audio mode: 0x00350035 Connect mode: 0x90000400 The command on the PRI gateway that causes the faxes not to send is this line under voice service voip: >> modem passthrough nse codec g711alaw redundancy which is what was added last week to get the modem transfer in the chip and pin machines to work. Using prserv to debug the call connection process from the ATA is somewhat helpful, showing that when the modem passthrough command is turned off on the PRI gateway, the ATA sends a V21 Flag after the dist packet. The traces show that the codec is g711ulaw and the NSE mode is 100: A working trace is: <-|StOffHk[1] |->SetRingerMode[1]:1 |->CallState[1]=1 L=1 R=2664e0a FG=0 r=0,0 FG NewState[1]=1, L=1, R=2664e0a [1]StopTone [1]StartTone 0 [1]DTMF 7 , insum:683243 [1]StopTone [1]StopTone [1]DTMF 2 , insum:644452 [1]DTMF 0 , insum:667693 [1]DTMF 8 , insum:656706 [1]DTMF 0 , insum:663984 [1]DTMF 8 , insum:653994 [1]DTMF 4 , insum:649581 [1]DTMF 3 , insum:636602 [1]DTMF 5 , insum:646275 [1]DTMF 2 , insum:662297 [1]DTMF 4 , insum:614936 [1]DTMF 0 , insum:658355 [1]DTMF 9 , insum:659060 [1]DTMF 9 , insum:642789 [1]DTMF 6 , insum:649316 |->DialedNumberMsg[1]:"720808435240996" |->CallState[1]=12 L=1 R=2664e0a FG=0 r=2664e0a,0 |->CallInfo[1] |->CallState[1]=3 L=1 R=2664e0a FG=0 r=2664e0a,0 |->CallInfo[1] |->CallInfo[1] [1]StopTone |->OpenRxChn[1] [1]StopTone <-|OpenRxChnAck[1] Save Media[1] |->StartMediaTx[1]:pt=0 ps=20 ss=0 ip=a1f0142:18604 [1]MPT mode 0 [3]MPT mode 0 Enable LEC adapt [1]=1 [1]Enable encoder 0 |->CallState[1]=5 L=1 R=2664e0a FG=0 r=2664e0a,0 |->CallInfo[1] [1]StopTone |->CallInfo[1] |->CallInfo[1] [1]DPKT 1st: 870500240 870500080, pt 0 V21 Flag [1]Tx MPT PT=100 NSE pkt c0000000 [1]MPT mode 2 [1]codec: 0 => 0 [1]MPT mode 3 988:00;0,6,0,0, <-|EndCall[1] <-|SK[1]:9, L=1, FGR=2664e0a [1]StopTone ->CloseRxChn[1] |->StopMediaTx[1] |->CallState[1]=2 L=1 R=2664e0a FG=0 r=2664e0a,0 IDLE[1] |->SetRingerMode[1]:1 And a non-working trace with the modem passthrough command removed on the PRI gateway is below. Interestingly this shows the line "[1]Fax Pass Thru RX Rdn" which is suggesting that when modem passthrough is on that fax transmission is trying to use fax passthrough. <-|StOffHk[1] |->SetRingerMode[1]:1 |->CallState[1]=1 L=1 R=2664e32 FG=0 r=0,0 FG NewState[1]=1, L=1, R=2664e32 [1]StopTone [1]StartTone 0 [1]DTMF 7 , insum:690242 [1]StopTone [1]StopTone [1]DTMF 2 , insum:652655 [1]DTMF 0 , insum:659085 [1]DTMF 8 , insum:653420 [1]DTMF 0 , insum:661407 [1]DTMF 8 , insum:683754 [1]DTMF 4 , insum:619270 [1]DTMF 3 , insum:651621 [1]DTMF 5 , insum:675484 [1]DTMF 2 , insum:660054 [1]DTMF 4 , insum:650734 [1]DTMF 0 , insum:669983 [1]DTMF 9 , insum:646790 [1]DTMF 9 , insum:637777 [1]DTMF 6 , insum:648348 |->DialedNumberMsg[1]:"720808435240996" |->CallState[1]=12 L=1 R=2664e32 FG=0 r=2664e32,0 |->CallInfo[1] |->CallState[1]=3 L=1 R=2664e32 FG=0 r=2664e32,0 |->CallInfo[1] |->CallInfo[1] [1]StopTone |->OpenRxChn[1] [1]StopTone <-|OpenRxChnAck[1] Save Media[1] |->StartMediaTx[1]:pt=0 ps=20 ss=0 ip=a1f0142:18526 [1]MPT mode 0 [3]MPT mode 0 Enable LEC adapt [1]=1 [1]Enable encoder 0 |->CallState[1]=5 L=1 R=2664e32 FG=0 r=2664e32,0 |->CallInfo[1] [1]StopTone |->CallInfo[1] [1]DPKT 1st: 3556639503 3556639343, pt 0 |->CallInfo[1] 991:00;0,2,0,0, [1]Rx MPT PT=100 NSE pkt c0000000 [1]Tx MPT PT=100 NSE pkt c0000000 [1]MPT mode 1 [1]codec: 0 => 0 [1]Tx MPT PT=100 NSE pkt c0000000 [1]MPT mode 2 [1]codec: 0 => 0 [1]Rx MPT PT=100 NSE pkt c0000000 [1]MPT mode 3 [1]Rx MPT PT=100 NSE pkt c0000000 [1]Fax Pass Thru RX Rdn [1]codec: 0 => 8 [1]MPT mode 4 estat[0]=17541402,96517,5134,0,0,0,E05FB9EE3E89,000000000000 edif=102295,499,0,0,0,0 estat[0]=17643704,97017,5134,0,0,0,E05FB9EE3E89,000000000000 edif=102302,500,0,0,0,0 estat[0]=17746166,97518,5134,0,0,0,E05FB9EE3E89,000000000000 edif=102462,501,0,0,0,0 estat[0]=17848416,98017,5134,0,0,0,E05FB9EE3E89,000000000000 edif=102250,499,0,0,0,0 991:30;0,6,0,0, ->CloseRxChn[1] |->StopMediaTx[1] [1]StopTone [1]StopTone [1]StartTone 2 992:00;0,6,0,0, <-|EndCall[1] <-|SK[1]:9, L=1, FGR=2664e32 [1]StopTone |->CallState[1]=2 L=1 R=2664e32 FG=0 r=2664e32,0 IDLE[1] |->SetRingerMode[1]:1 I'd like to be able to have modem passthrough enabled on the PRI gateway to allow chip and pin machines to negotiate properly, but to have fax machines not use passthrough. I've tried setting the fax passthrough to prefer nse then fall back to cisco propriatary mode on the PRI gateway, and this has no effect. Has anybody ever seen this or a similar issue before, and are you able to help? Thanks, Matt

weavermj1982 · ‎07-13-2012

Thanks Paolo. I've managed to get modem passthrough to work. Modem passthrough is not actually supported with SCCP at all. I had to set modem passthrough globally at the voice service voip level on the VG224, and also on a new voip dial peer which points to my PRI gateway. On the PRI gateway I also had to configure modem passthrough at the voice service voip level globally. on the VG224: voice service voip modem passthrough nse codec g711alaw redundancy dial-peer voice 1 voip preference 1 destination-pattern 8T modem passthrough nse codec g711alaw redundancy session target ipv4:10.1.1.1 codec g711ulaw and on the PRI gateway: voice service voip modem passthrough nse codec g711alaw redundancy Obviously this doesn't have any class of restriction on the lines, but it is sufficient for us to perform a workaround in the short-term. Thanks for the initial pointer RE SCCP :-) Matt

weavermj1982 · ‎07-12-2012

pri_gw#sh controllers e1 E1 0/0/0 is up. Applique type is Channelized E1 - balanced Cablelength is Unknown Description: PSTN No alarms detected. alarm-trigger is not set Version info Firmware: 20090408, FPGA: 13, spm_count = 0 Framing is CRC4, Line Code is HDB3, Clock Source is Line. Data in current interval (730 seconds elapsed): 0 Line Code Violations, 0 Path Code Violations 0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins 0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs Total Data (last 24 hours) 0 Line Code Violations, 0 Path Code Violations, 0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins, 0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs Thanks Paolo, I was considering setting up modem relay but it always seems to look more complicated that it needs to be. That will be my next thing to try. Have you had bad experience with modems working over SCCP connections then? No slips on the E1 interface so looks like the SIP trunk is the next thing to do.

weavermj1982 · ‎07-12-2012

Hi fellow community members, I have a very specific problem that I'd appreciate some help with if anyone else has experienced this or anything similar. We have a VG224 that provides analogue lines to mainly fax machines on our campus. Recently we have had our finance department want to use chip and pin machines over these connections. Previoulsy we have used streamline machines, and these connect fine. The specific case I have is a Verifone vx810 machine that is connected to a VG224. The Verifone unitis able to dial out and successfully authorise a transaction, but it cannot complete a batch upload process or a TMS. I've spoken to the support company who we lease the machines from and they have identified that the machine uses the following baud rate, parity and stop bits for the 2 different transactions: For the transactions: 2400 baud, parity/stop 7e1 (this works) For the batch upload/TMS : 19200 baud, parity/top 8n1 (this doesn't work). In the case of the batch upload/TMS, the machine dials out, obtains a connection to the remote end over the PSTN for about 10-20 seconds, then tears it down because the modem negotiation fails. I've read various articles on here and other forums saying that high speed data rates may be an issue for the VG224, but nothing that suggests a problem with 19200. I also tried the setup on an ATA186 and the ATA showed the same symptoms, so I'm inclined to think this isn't just a firmware or bug issue with the VG224. I've tried 3 different chip and pin machines of the vx810 model and all of them show the same symptoms. Here is a copy of my current VG224 config. The VG224 is registered to the CUCM via SCCP. version 12.4 no service pad service timestamps debug datetime localtime service timestamps log datetime localtime no service password-encryption ! hostname vg224 ! boot-start-marker boot-end-marker ! logging message-counter syslog logging buffered 4096 ! aaa new-model ! ! aaa authentication login default group radius local aaa authentication enable default enable aaa authorization exec default group radius local aaa accounting exec default action-type start-stop group radius ! ! ! aaa session-id common clock timezone GMT 0 clock summer-time BST recurring 4 Sun Mar 1:00 4 Sun Oct 2:00 ip source-route ip cef no ip domain lookup ! ! no ipv6 cef ! stcapp ccm-group 1 stcapp ! stcapp feature access-code ! stcapp feature speed-dial ! ! voice service voip modem passthrough nse codec g711ulaw ! ! voice-card 0 ! username <username> password <password> archive log config hidekeys ! ! ! interface FastEthernet0/0 no ip address duplex auto speed auto ! interface FastEthernet0/0.644 encapsulation dot1Q 644 ip address 10.1.160.4 255.255.255.0 ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! ip default-gateway 10.1.160.1 ! ip forward-protocol nd ip route 0.0.0.0 0.0.0.0 10.1.160.1 no ip http server ! logging facility local6 logging source-interface FastEthernet0/0.644 logging <server ip> snmp-server community <community> RO snmp-server location <location> ! radius-server host <server> auth-port 1812 acct-port 1813 radius-server host <server> auth-port 1812 acct-port 1813 radius-server timeout 3 radius-server key <key> ! control-plane ! ! ! voice-port 2/0 cptone GB timeouts initial 60 timeouts interdigit 60 timeouts ringing infinity caller-id enable ! voice-port 2/1 cptone GB timeouts initial 60 timeouts interdigit 60 timeouts ringing infinity caller-id enable ! voice-port 2/2 cptone GB timeouts initial 60 timeouts interdigit 60 timeouts ringing infinity caller-id enable ! voice-port 2/3 cptone GB timeouts initial 60 timeouts interdigit 60 timeouts ringing infinity caller-id enable ! voice-port 2/4 no echo-cancel enable cptone GB timeouts initial 60 timeouts interdigit 60 timeouts ringing infinity caller-id enable ! ! ccm-manager fax protocol cisco ccm-manager config server <publisher> ccm-manager config ccm-manager sccp local FastEthernet0/0.644 ccm-manager sccp ! ! sccp local FastEthernet0/0.644 sccp ccm <pub>identifier 1 version 6.0 sccp ccm <sub1> identifier 2 version 6.0 sccp ccm <sub2> identifier 3 version 6.0 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate ccm 2 priority 2 associate ccm 3 priority 3 ! dspfarm profile 1 transcode associate application SCCP ! ! dial-peer voice 999200 pots service stcapp port 2/0 ! dial-peer voice 999201 pots service stcapp port 2/1 ! dial-peer voice 999202 pots service stcapp port 2/2 ! dial-peer voice 999203 pots service stcapp port 2/3 ! dial-peer voice 999204 pots service stcapp port 2/4 ! ! ! line con 0 line aux 0 line vty 0 4 ! ntp server <ntp1> ntp server <ntp2> end And a show version from the vg224: System returned to ROM by power-on System restarted at 14:30:34 BST Wed May 9 2012 System image file is "slot0:vg224-i6s-mz.124-22.T5.bin" Cisco VG224 (R527x) processor (revision 4.1) with 119808K/11264K bytes of memory. Processor board ID FHK1432F2CC R527x CPU at 225MHz, Implementation 40, Rev 3.1 1 On-Board Twenty-Four FXS Analog Voice Module V2.1 2 FastEthernet interfaces DRAM configuration is 64 bits wide with parity disabled. 63K bytes of non-volatile configuration memory. System fpga version is 250027 System readonly fpga version is 250027 Option for system fpga is 'system'. 62496K bytes of ATA Slot0 CompactFlash (Read/Write) Configuration register is 0x2102 This problem is really driving me mad, if anyone can shed any light on what the root cause may be I'd be very grateful. I should probably add that yes, PCIDSS compliant devices running over the IP network would be the way forward, and that is something we are working towards, but currently we have units that can only communicate using analogue telephone lines.

weavermj1982 · ‎03-16-2012

I had a similar problem last week doing a factory reset on a 7942 phone running SCCP42.9-1-1SR1S on a Call Manager Express 8.6 setup. Within CME the phone would just keep looking for term42.default and sit their loooping. To overcome the issue I connected the phone to a CUCM 6.1 install and it successfully uplaoded the default boot loader and flashed to an 8.0 SCCP image - then I connected the phone back to CME and it upgraded back to 9.1 Hope this helps - maybe it is your tftp bindings within CUCM? Do you have access to another CUCM system that you can try to firmware it from?

weavermj1982 · ‎02-15-2011

Did anyone find a resolution to this bug? I need to bulk add services to around 800 phones.

weavermj1982 · ‎02-09-2011

Tarik, I finally got around to implementing this today and can confirm that I now have call preservation working across our cluster. Thank you for your pointers as this is an issue that we have been wanting to fix for a long time. Best wishes, Matt

weavermj1982 · ‎12-09-2010

Thanks Tarik, sadly this did not work and the audio dropped on the outbound call leg 15 seconds after the connection to the publisher (primary CM) was lost. Do you have any expertise/experience in this area? I'd really like to get outbound call preservation working for piece of mind should any 999 calls be routed out to the PSTN during a failover period. Matt

weavermj1982 · ‎01-14-2010

Hi Joseph, We are running a CUCM 6.1 install with one pub and 2 subs. We have the same issue whereby in progress calls from an IP phone registered to a sub as the active CM drop around 15 seconds after the subscriber is failed over to the pub. Did you ever get a resolution to this problem? I'd be interested to hear from others who have the same problem, as I don't think it's right that active calls should be dropped. Thanks, Matt

weavermj1982 · ‎07-20-2007

We have a cisco 3725 gateway adn we have just started to monitor our PRI utilisation with SNMP. This OID will return the number of B channels in use for each serial interface in the gateway: .1.3.6.1.4.1.9.10.19.1.1.9.1.3.1 To get the stats for individual interfaces, add a .0, .1 etc for each interface as follows: .1.3.6.1.4.1.9.10.19.1.1.9.1.3.1.0 = stats for first serial interface .1.3.6.1.4.1.9.10.19.1.1.9.1.3.1.1 = stats for second serial interface You can log these values to a text file in CSV format or graph them with something like Cricket. In our example this gives us the utilisation of our PRI to the PSTN and a PRI that we have to our PBX You could also use a script like this to get the values: #!/bin/bash #Declare a date variable of the format YYYYMMDD date=`date +"%Y-%m-%d"` time=`date +"%T"` pstn=`snmpwalk -c -v2c .1.3.6.1.4.1.9.10.19.1.1.9.1.3.1.0 | awk '{print $4}'` pbx=`snmpwalk -c -v2c .1.3.6.1.4.1.9.10.19.1.1.9.1.3.1.1 | awk '{print $4}'` if [ "$pstn" == "" ] then pstn='0' fi if [ "$pbx" == "" ] then pbx='0' fi echo $date $time,$pstn,$pbx >> stats.txt Cron this script every minute.

weavermj1982 · 07-18-2012

Hello fellow support forum members,Last week I posted an issue I was having where some Verifone chip...

weavermj1982 · 07-13-2012

Thanks Paolo.I've managed to get modem passthrough to work. Modem passthrough is not actually suppor...

weavermj1982 · 07-12-2012

pri_gw#sh controllers e1E1 0/0/0 is up. Applique type is Channelized E1 - balanced Cablelength is Un...

weavermj1982 · 07-12-2012

Hi fellow community members,I have a very specific problem that I'd appreciate some help with if any...

weavermj1982 · 03-16-2012

I had a similar problem last week doing a factory reset on a 7942 phone running SCCP42.9-1-1SR1S on ...

weavermj1982 · 05-20-2011

Hi Andy,This thread is really useful to me. We have a 6.1(2)-1000-13 install (1 pub and 2 subs) that...

weavermj1982 · 02-15-2011

Did anyone find a resolution to this bug? I need to bulk add services to around 800 phones.

weavermj1982 · 02-09-2011

Tarik,I finally got around to implementing this today and can confirm that I now have call preservat...

weavermj1982 · 12-09-2010

Thanks Tarik, sadly this did not work and the audio dropped on the outbound call leg 15 seconds afte...

weavermj1982 · 01-14-2010

Hi Joseph,We are running a CUCM 6.1 install with one pub and 2 subs. We have the same issue whereby ...

weavermj1982 · 07-20-2007

We have a cisco 3725 gateway adn we have just started to monitor our PRI utilisation with SNMP.This ...

Date Registered	‎07-20-2007 07:43 AM
Date Last Visited	‎08-18-2017 03:56 AM
Total Messages Posted	12
Total Helpful Votes Received	4

ATA186 fax issue when modem passthrough...

Re: VG224 and Verifone xx810 chip and p...

VG224 and Verifone xx810 chip and pin t...

VG224 and Verifone xx810 chip and pin t...

hard reset killed cisco 7942

Re: Bulk Subscribed Phone Services CUCM...

Re: External calls drop during server f...

Re: External calls drop during server f...

Re: External calls drop during server f...

Re: PRI voice utilization

ATA186 fax issue when modem passthrough enabled on PRI gateway

Re: VG224 and Verifone xx810 chip and pin terminal modem diallin

VG224 and Verifone xx810 chip and pin terminal modem dialling

VG224 and Verifone xx810 chip and pin terminal modem dialling

hard reset killed cisco 7942

Re: Upgrade to CUCM 6.1(5) from 6.1x or 6.2x possible?

Re: Bulk Subscribed Phone Services CUCM 6.x

Re: External calls drop during server failover

Re: External calls drop during server failover

Re: External calls drop during server failover

Re: PRI voice utilization