Multi-FXO IP ATA?

I've actually installed this before on Corporate jobs. I just assumed they were way too expensive for this project. I'll go ahead and see if I can find some 4 Port VWICs that will do what I'm looking for. I'm a CCNA so hopefully I won't have too much of a nightmare configuring them.

IF anyone else has any other suggestions feel free to chime in. As reliable as Cisco hardware is, it has it's quirks.

Greenwire IT
Personal Site

I think you sold me Scott. 150$ for a Four Port Cisco Gateway and 4 FXO ports is irresistible. I'll go ahead and order one for testing.

Thanks again!

I did forget to tell you one thing, most 1751's shipped with the PVDM (DSP) DIMM, if this one did not you will have to hunt up a PVDM on eBay. If you need one in a jiffy PM me I have a bunch of them I can harvest off of NM-HDV that I have pulled out of service.

Scott

I offered 20$ shipped for a 12 Channel on eBay, and the guy accepted it. Foresee any problems with this one, http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=290293838680 ?

I noticed it says PVDM2 instead of PVDM, any idea what the difference is?

I've love to, I know there are other people in my position, and this project is going to cost me less than a Linksys SPA400. That's a steal.

Paging Scott 'SkykingOH' to this thread! I sent a message to the guy I bought the DSP from, no response. I really hope I didn't actually buy the SIMM version. If so, 20$ isn't a catastrophe.

Damn, that's alright, I'll list it for 25$ buy it now with free shipping and break even :D

Take a look see at this one: http://cgi.ebay.com/CISCO-PVDM-12-12-Channel-Packet-Voice-Fax-NM-HDV_W0QQitemZ300318514619QQcmdZViewItemQQptZCOMP_EN_Routers?hash=item45ec60ddbb&_trksid=p3286.c0.m14&_trkparms=72:570|66:2|65:12|39:1|240:1318|301:0|293:1|294:50

Yeah - those are the right ones. You made me work, I only posted auction ID's you posted the whole screwed up URL.

Look like a pro, get the BBCode plug in for Firefox. format your posts easy.

So the router already had a PVDM? That is why I suggested you wait.

I have not upgraded anything I have to 12.4 yet, too many things have changed. Here is the build I run my gateways on:

I also included a working Asterisk Dial Peer. It's for a PRI however it should get you pointed in the right direction I sanitized incoming called number field, you don't need it, I use it to redirect DID's to different serves.

MAI_PRI_Gateway#sh ve
Cisco Internetwork Operating System Software
IOS (tm) 3600 Software (C3620-IS-M), Version 12.2(13)T9,  RELEASE SOFTWARE (fc2)
TAC Support: <a href="http://www.cisco.com/tac" title="http://www.cisco.com/tac">http://www.cisco.com/tac</a>
Copyright (c) 1986-2003 by cisco Systems, Inc.
Compiled Thu 02-Oct-03 10:54 by pwade
Image text-base: 0x6000891C, data-base: 0x61850000

ROM: System Bootstrap, Version 11.1(7)AX [kuong (7)AX], EARLY DEPLOYMENT RELEASE SOFTWARE (fc2)

MAI_PRI_Gateway uptime is 30 weeks, 6 days, 3 hours, 20 minutes
System returned to ROM by power-on
System image file is "flash:c3620-is-mz.122-13.T9.bin"

cisco 3620 (R4700) processor (revision 0x81) with 53248K/12288K bytes of memory.
Processor board ID 03063454
R4700 CPU at 80Mhz, Implementation 33, Rev 1.0
Bridging software.
X.25 software, Version 3.0.0.
Primary Rate ISDN software, Version 1.1.
1 FastEthernet/IEEE 802.3 interface(s)
25 Serial network interface(s)
1 Channelized T1/PRI port(s)
DRAM configuration is 32 bits wide with parity disabled.
29K bytes of non-volatile configuration memory.
16384K bytes of processor board System flash (Read/Write)

Configuration register is 0x2102

dial-peer voice 100 pots
 preference 2
 destination-pattern .T
 direct-inward-dial
 port 0/0:23
 forward-digits all
!
dial-peer voice 2 voip
 description Route Calls to Asterisk
 preference 1
 incoming called-number 216xxxxxxx
 destination-pattern .T
 session protocol sipv2
 session target ipv4:192.168.10.252
 session transport udp
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
gateway
!
sip-ua
 retry invite 2
 retry response 3
 retry bye 3
 retry prack 6
 timers expires 300000
 sip-server ipv4:192.168.10.252
 no transport tcp
!
!

The config I sent you forwards caller-id

You mean name data? Never tried I do that downstream.

PVDM arrived, no PVDM light, so I did some further inspecting. sh diag did not see the PVDM or the VIC listed. At first I thought I would need a Voice IOS, but unfortunately it looks like both of the PVDMs I ordered are not correct. The 1751 apparently requires the PVDM-256K. I now have a PVDM2-12 and a PVDM-12 both probably worth something for someone else, but not much use for my 1751.

http://www.cisco.com/en/US/products/hw/routers/ps221/products_dat...

Any advice? I would jump back on eBay for a third go, but it looks like most of the reasonably priced PVDMs are in China or Hong Kong, and I'd rather not wait three weeks for a PVDM. If you have any of these around I'd be more than happy to reimburse you for your trouble.

I'm going to try and score Item number: 140325866039 but it's still two days out. I sent some messages to the Chinese sellers looking for estimates on arrival times. We'll see what they say.

Bought a PVDM-256K-4 for 33 shipped from HK. Guy said he ships 2/3 day DHL. I'll let you know how it turns out.

Thanks anyway for the advice, as soon as I get this PVDM problem out of the way it's going to be interesting writing a guide for these.

In theory, you could use a cisco router with a couple bri interfaces as well?

Up in canada isdn was pretty popular and t1s are still pretty expensive. I'm thinking this might sold some problems for me as the diva cards aren't so easy to configure!

I got a little crazy and I bought this 1760 as well: 200349335345

haha, I thought that was particularly precious when I saw it said (4x) Ethernet. I was debating sending him a complaint when it arrived that it in fact only had 1FE instead of the 4ESW I was promised. Or maybe complaining that the RJ48X/Console port doesn't work correctly with my Ethernet switch. After stealing it for 100$ though I don't think I could bring myself to that. I am thinking about picking up one of those 4ESWs for around 50$ and using it to get rid of the 5pt in my wiring closet.

I don't think I've ever purchased a single piece of Cisco hardware first hand. It's just not cost-practical most of the time. I really wish it was because it'd be a lot easier to standardise builds and hardware. I guess I'll have to just be content that each IP Gateway I build will be slightly different!

I know you mentioned you could use some PVDM-256Ks, here is one that might go for pretty cheap: 140325866039 put in a snipe for one of those 1751s you have laying around.

Alright, that's it. I bought my last piece 110399395745 . I'm done with eBay for a while. The whole router with the MOD-VPN, the 4ESW and Modem was cheaper than just the WIC-4ESW alone. I'm going to cannabalise it.

Here's my total haul
$60 Cisco 1711 w VPN-MOD, Modem, 4ESW 96DRAM
$50 + $40 + $30Cisco 1751 w VPN-MOD, 2FXO, PVDM-256K-4
$120 Cisco 1760 w 2FXO T1 CSU/DSU
Grand Total $300.
That's assuming I can sell my two PVDMs for 30$/pop which I don't imagine being a problem.

That means for 300$ I got:

Cisco 1751 w/ VPN / 2FXO / PVDM / 96DRAM / T1 DSU/CSU
Cisco 1760 w/ VPN / 2FXO / PVDM / 96DRAM / T1 DSU/CSU / Modem / 4Port Switch

I don't think I can imagine a better lab environment to play with. Having these two VPN modules means that I should be able to play with doing SIP over encrypted VPN.

Do you know if you can configure the FE ports on the 4ESW as normal fe routing interfaces or will they only function as switch ports?

Excellent advice. I don't plan on using them for anything except as a VoIP lab, although I'd like to push them in terms of the number of features I can use (QoS/VLAN/IPSec) simultaniously. So I can transfer these skills to a production environment down the road. I don't plan on using more than one of the switch ports for a routing interface.

I looked up some information on specific pps routing abilities. The 1751 can do 12000pps and the 1760 can do 16000 (No ACL/Route Maps). So you were right on. I've always been a bit curious about this. Does MTU much effect how many pps a router can process? With a MTU of 1500bytes, that comes out to 15,000 x 1500 = 22,500kbytes/sec or 180mbps. However at 64byte packets that comes out to 7.68mbps. I guess it boils down to, what kind of actual throughput do you think I should expect from these, assuming no QoS/ACL/etc.

That's very interesting about GRE, I'll have to play with that, having QoS control over the tunnel would be helpful. Some applications require encryption on public networks (HIPAA) so it'd be neat to be familiar with SIP over IPsec. In the past I've used ssh or stunnel.

I've read that you can pass GRE over an IPsec VPN. Theoretically couldn't you establish a GRE connection over an IPSec tunnel and then use QoS within the GRE interface?

One missed question:

I guess it boils down to, what kind of actual throughput do you think I should expect from these, assuming no QoS/ACL/etc?

I'm going to start writing the article in a little bit. At least a preliminary section explaining the hardware (VICs/PVDMs) etc.

VIC2's are newer and would require a far more expensive router. There is not a functional difference.

A 2610 with a NM-2V and 2 dual FXO cards would be a very cheap rack mount setup. We are working on a wiki entry with complete details. You should run IOS 12.2 anything newer will take quite a bit more RAM and FLASH.

Maxing out the Memory (and flash) is never a bad thing.

Excellent, let me know how the trunk building goes. Inbound is a no brainer, I should have posted this also....trunk config.

host=192.168.10.192
username=mg1
disallow=all
allow=ulaw
type=peer
dtmf=rfc2833
insecure=very
canreinvite=no
context=from-trunk
qualify=yes

The username is unneeded however it prints in the CDR so I add it. You don't need an entry in the inbound side this will take care of both.

The entering some numbers to complete the call makes sense. I send you a config for DID's. Since POTS doesn't feed numbers inband I am sure what you are entering appears as the CID.

I have a cookout tonight or I would go digging.

FXO's need to be in PLAR mode if memory serves me correct to forward. You also need to have caller-id enabled on the physical port.

That information is old and sketchy on voip-info.

Battery reversal is not part of caller-ID.

The standard modules support Bell103a style caller-ID presentation.

I want to get your inbound working though. Have you debugged on the Cisco? When you dial with the plar statement does it send out a SIP invite?

A guide would be most welcome indeed.

I had found through Google an old (almost a year) post from Scott talking of success with the 1750 and PM'ed him. He kindly pointed me to this thread. (Thanks SkykingOH!)

I've had a system with one SPA3000 and one PAP2 running for over half a year, but this is a completely different animal. I'd never been thrilled with the call quality of the SPA/PAP2, but it was "good enough" for my meager needs.

The SPA3000 blew up, literally. I get a green light but nothing else, no web interface, not even the fail-over relay clicks when power is applied like it used to...

I ended up on ebay. After some searching and multiple bids I built myself a 1750 with one VIC-2FXS and one VIC-2FXO-M1 (for caller id), PVDM-12, 48MB DRAM and 16MB Flash, IOS IP/Voice Plus firmware 1.23(26).

It seems that most of the "normal" SIP stuff (registration/authentication - what I'm used to with the SPA and PAP2) found its way in to IOS after they discontinued support of the 1750, but reading through the configurations posted here I'm not seeing that used so I'm hopeful I can get this up and running.

I'm still trying to wrap my head around the interactions in the configuration. I understand the voice-port command configures the physical card/port, and that the dial-peer command (somehow) sets up the associations but that's where it all starts to unravel in my head. I'm about as far from a CCNA/CCNE as you can get and still be able to understand the basic workings of an IP-PBX. :)

One basic question up front that might help me understand the configurations better... Do the numerical tags like the 100 in "dial-peer voice 100 pots" or the 2 in "dial-peer voice 2 voip" mean anything to the TB settings? Or are the TB "settings" somehow specified in the destination-pattern (or other variables under the dial-peer group)?

I've already downloaded the VoIP Quick Start Guide and the IOS Voice Command reference from Cisco (which is just shy of 3000 pages!) as cures for insomnia. Any help you can provide would be great.

Scott,

I hate to think of it as "leaching", I would like to try and learn how it all goes together rather than just copying something verbatim for a post off the forums.

I'll look into picking up a Cisco gateway reference (any suggestions?) as well as trying to spend some additional time with the two I already found on the Cisco website.

Moving a bit off the FXO subject, is setting up an FXS much different? Ideally I'd like to retire the PAP2 and get the VIC-2FXS running as well.

Thanks for the pointer to voip-info, I'll look that up as well.

Mark

BTW - Don't worry about aggravating me (not that you would). I'm doing this in between my day job, sifting through pictures from two dance recitals and working on an engineering consulting job for an RF transmitter. Oh yeah, I also have three young girls that demand almost constant attention. I understand busy. :)

Scott,

I appreciate the link. I've already been there, done that, no love earlier in the week.

The problem lies in the authentication line... Here is what I get from my 1750.

cisco1750(config)#voice-port 1/0
cisco1750(config-voiceport)#mwi
                             ^
% Invalid input detected at '^' marker.

cisco1750(config-voiceport)#station-id number 1001
cisco1750(config-voiceport)#exit
cisco1750(config)#dial-peer voice 1001 pots
cisco1750(config-dial-peer)#destination-pattern 1001
cisco1750(config-dial-peer)#direct-inward-dial
cisco1750(config-dial-peer)#port 1/0
cisco1750(config-dial-peer)#authentication username 160 password 160
                             ^
% Invalid input detected at '^' marker.

cisco1750(config-dial-peer)#

The only thing I changed was the physical port assignment. My FXS is in Slot 1, so the ports are 1/0 and 1/1. I'm not too concerned about the mwi error in the port setup, message waiting indication support is not critical.

What I find truly odd is that the firmware isn't that old (Compiled Mon 17-Mar-08 14:24 by dchih), so I would expect it to work given that the thread you linked is from 2006.

Either the 1750 does not support SIP authentication (hard to believe, but not impossible) or there is something that needs to be enabled/configured first. Since you had posted success with the 1750 (a year ago I realize) I had great hope.

Further confusing the entire issue is trying to decipher Cisco's revision numbering scheme. According to The June 2007 IOS Voice Command Reference, "mwi" was added as of IOS 12.3(8)T so why wouldn't I have it in 12.3(26) which was released in March 2008?

I really appreciate your help, this has been frustrating me for almost a week already.

Mark

Scott,

Yes, there is a public (or so it seems) version of the IOS feature tool that lets me search based on platform, IOS revision and features.

I am running the latest IP/VOICE PLUS version available for the 1750. What feature or command should I be looking for?

Is there any way you can find out which IOS you were running on the 1750's that you had working? (Unlikely I know, but I had to ask).

Thanks

Mark

Here is a sample config for Cisco FXS that works for me.

voice-port 1/0
cptone GB
caller-id alerting line-reversal
!
dial-peer voice 1000 pots
destination-pattern 100
port 1/0
authentication username 100 password sippassword
!
sip-ua
retry invite 1
registrar ipv4:192.168.1.12 expires 3600
!

The unit on my bench is a 1751 (just had a guy look I was talking to), I have to look on the Cisco web site and see what the difference is.

OMG!

(Sorry for the teenage-ism)

I have successfully set up both FXS ports and it appears the FXO port is active (I'm not testing any further @ 12:15am, the wife would kill me). I've placed calls (ring verification only) both ways between an existing extension on the PAP2 and one of the two FXS ports on the 1750.

The only thing I'm not thrilled about is the lack of a "secret" for the connections, but since the 1750 is on a (private) static IP I'm not too worried. (Should I be?)

Scott, I appreciate the time you took to assist me. Once I've had a chance to verify everything is working (no one-way audio issues, etc) I'll post up my settings for any who dare sail these waters in the future.

Going to bed tonight without Cisco indigestion.

Mark

OK, adding the following based on Scott's post above, and now incoming calls work!

The only remaining issue is I have to use the "catch-all" inbound route (no DID, no CID).

Is there a way to get it in to a specific DID group? Could it be related to the FXO station-id=#1 and the DID not allowing "#"?

dial-peer voice 100 voip
 preference 1                              <-- New
 incoming called-number .
 destination-pattern .T                <- Added "." in front of T
 voice-class codec 10
 session protocol sipv2
 session target ipv4:192.168.0.10:5060
 session transport udp                               <-- New
 dtmf-relay rtp-nte
 no vad

Just so I understand, you think the "300" in ipv4:300@192.168.0.10:5060 will be interpreted by asterisk as the DID?

Thanks again for the reply, I'll give it a shot in a bit.

Scott, no luck. Guess I'll just live with the Any/Any catch-all.

Let me know if you have any other suggestions.

cisco1750(config-dial-peer)#session target ipv4:300@192.168.x.x:5060
Incorrect format for Session Target
Must be of the form     ^((loopback:rtp)|(dns:.*)|(ipv4:[0-9]+\.[0-9]+\.[0-9]+\.
[0-9]+(:[0-9]+)?)|(enum:([1-9]|1[0-5]))|(ras)|(sip-server))$

and

cisco1750(config)#voice class sip-profiles 1
                              ^
% Invalid input detected at '^' marker.

cisco1750(config)#voice class ?
  aaa                     AAA Parameters
  busyout                 Set global voiceport busyout monitoring
  cause-code              Cause code list parameters
  codec                   Set codec global parameters
  custom-cptone           configuration of customized Call Progress Tones
  dualtone-detect-params  dualtone detection paramaters
  h323                    H323 Parameters
  permanent               Permanent connection (call)