First off I’d like to say that this is just a minor issue, more relating for routers versus switch, I’m still a lot happier at how Cisco implements config and features as opposed to most if not all of their competitors…

At a customer I’ve recently had to commit a grave operational sin; to connect a small switch at the end of a floor patch. These things are normally operational nightmares as they have a tendency to quickly bring an entire LAN environment down to its knees when such a ‘switch’ is connected to the network twice. Always by accident but having management kick you for something someone else did is not anyone’s idea of fun. I won’t go into the underlying principles here as I’m assuming most who frequent my blog will know about broadcast storms, their causes and the tools and solutions available to mitigate the risks.

Our justification to operations was that we wanted a few more local LAN ports to test VoIP devices on than we had available through floor patches. As such I reasoned with Operations that this was a calculated choice to segregate our testing from the rest of the LAN but still make it as realistic as possible. Using the means available meant that I had to make do with a Cisco 1801. Single routed and 8 switched interfaces. Think of it as a router with one Ethernet interface and an 8 port HWIC-ESW nailed to it. Didn’t need the ATM or WiFi it has.

So I set out, disabling IP routing, admin down all non-Ethernet ports. set up the vlan database -old style, remember?-; I did not want this baby to participate in VTP, in fact I don’t think it even can! It’s limited to 8 vlans. Pulled two cables to it. One switched port as trunked with some data and voice vlans and configured the routed interface for management access.

All sweet and dandy, tested the BPDU-guard functionality prior to installation by connecting an access-port to the LAN. Clunk! it went down as desired, result I thought… Then when installing the LAN wouldn’t bring up the LAN port. Doh! I’d missed that the 1801 doesn’t send BPDU’s until a VLAN becomes active. I’d checked if spanning-tree was operational, and it wasn’t until I brought an interface up. So I disabled STP for all vlans in the VLAN database. Now my laptop received an IP address and the data VLANs all worked.

So, time to connect a Mitel phone. No dice, it received it’s first DHCP response with VLAn information, then it would just sit ennuncing it was waiting for a DHCP response. Dang, I’d configured the voice vlan so why did the switch not detect the phone, enable trunking so that the phone could send it’s DHCP request on the voice VLAN?

It was only when I started reading up on HWIC-ESW voice-VLAN config I noticed that Cisco hasn’t implemented the auto enable of dot1q trunking when a phone is detected… The solution is to add two lines of code; “switchport truck native vlan xyz” and “switchport mode trunk”. The crux is that this platform is at heart a router, not a native switch…

Cisco documentation

Categories: Networking, VoIP, Work
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For me SIP is the best option with most countries that I call being free and the UK being equal or cheaper than my current land line carrier. However my family in the UK has a better deal with Skype when the majority of the calls are national (1.4p per min and 4.5p per call). So if they have a stand alone Skype phone and I have SIP phones, should we not be able to call each other for free?

My options are:

1. PSGw requires an additional pc and single Skype account per pc
   
2. Skype Asterisk channel (chan_skype) Can only run on Asterisk but can run multiple instances of Skype through vnc

Not for me:

1. ‘Uplink Skype to SIP Adapter’ (windows required)
2. CooSIP, no idea about price and it’s not available yet
3. Skip2PBX, way too expensive
4. Pika Connect for Skype, way too expensive hardware & licenses [2009-01-07 no more references to Skype on their site]

chan_skype it will be…

Edit (23/05/2007): Well it’s end of May and I haven’t been able to spend more time on this yet, if you want to do this on Asterisk then no problem but on AsteriskNOW it’s not that easy.

Categories: VoIP
2 Comments »

Voipbuster is sporting free calls to quite a number of countries, sadly the UK isn’t one of them but at 1cpm I’m not complaining. However it’s good to realise one thing:

Voipbuster limits free call duration to one hour. After this hour the remote end get’s it’s call dropped while the local end is not notified. I presume this is to prevent automatic (re)dialing to these countries. One could circumvent this by static noise detection but that’s a little too advanced for most script kiddies.

Not an issue for me as I’ll just dial again when the remote end stops talking to me… 😉

Categories: VoIP
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After playing with the idea for a long time I decided it was time to implement a VoIP PBX. Reading up on the subject I found that I had 3 options which appealed to me: Asterisk (*) on Debian, AsteriskNOW (*NOW) or Trixbox. I opted for *NOW as it’s stripped bare and I hope I can add features as and when I need them. So far it’s gone well but I do have some additions I still need.

Installation of *NOW beta 4 (1.4.2) went fast on an Intel ISP2150 I had available, took about 30 minutes in total, including base config. Next I had to test it so I found a good free SIP soft-client in X-lite 3.0. And off I was phoning from laptop to laptop, surprisingly I had no issues with my WLAN.

The next move was to add two SIP accounts, Voipbuster and Xeloq. Both offer free national calls in NL additionally Voipbuster offers a range of free international calls while Xeloq has cheaper national mobile rates. Dialling from softphone to national landlines worked right away and the people I called didn’t even notice I wasn’t using my regular phone or line.

My server is now happily humming away in the datacenter and I have two 7960’s connected and registered across NAT, all is well so far. I’ve given some family members accounts and I’ve had a 2 hour conversation with my brother last weekend. We both have DSL and were quite happy with the quality of both the conversation and the connection. 🙂

Next my plan is to migrate my current landline number to a SIP carrier, not sure which one yet, and try to implement a SIP to Skype bridge. Further actions on my to do list are:

• Cancel my land line and save at least 18,50 Euro per month
• Get a (SIP) Siemens S450 IP dect phone to replace my current old Philips (pstn) dect
• Implement a phone book for the Cisco 7960 phones
• Implement some services for the Cisco 7960 phones
• Get a GSM SIP gateway and a mobile contract for backup/emergency calls and cost savings to national mobile numbers
• Implement a registry of calls (CDR?) so I can tell whether my bills will be correct
• Offer the SIP service to family members, using data from the previous point

Categories: VoIP
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