ramblings

Tag: Asterisk

Some info on MagicJack

by on Dec.27, 2010, under VoIP

Came across this interesting article on MagicJack at Digital Offensive. Got to try it out soon.

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Linksys SPA-3102

by on Sep.05, 2010, under VoIP

The SPA3102 that Kay uses was stuck in the off-hook state after an incoming call. This device is installed in a remote location he had to reboot it over the network.

Since it does not have “Reboot” button unlike most appliances, I asked Kay to browse to http://<spa.ip.address>/admin/reboot. That did the trick, he was thankful to yours truly, since it saved him a 3 hour trip!

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Asterisk and AVAYA

by on Aug.15, 2009, under VoIP

Some info I posted on ChipOverclock’s blog:

Thot i’d post my experiences with the AVAYA SIP phones.

  • The current version of the firmware for the 46xx phones still doesn’t seem to support the message waiting indicator (MWI) light. And as a side-effect of pending voice mail, these phones seem to think they are not registered after a seeming random period of time, refusing to allow outgoing calls. I had to disable voice mail on all extensions with AVAYA phones. Wasn’t too bad, I could live without the MWI.
  • The 1.xx version of SIP firmware for the 9620/9630/9640 series of phones would register fine on Asterisk, allow outgoing calls but refuse to accept incoming calls.
  • AVAYA have recently release version 2 of the SIP firmware for the 96xx phones. Thought I’d give it a try the other day. I was pleasantly surprised to find that it worked perfectly, including the MWI!! Excellent speaker-phone, and great voice quality.

I’ve been testing it for about 4 days now and am totally happy with the results.

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my experiments with Cisco 7940 IP phone

by on Jun.27, 2009, under VoIP

12:40 AM  09 Dec 2007
Seem to have figured out the strange and seemingly erratic behaviour of the Cisco 7940.

Quick answer… in the SIP<mac_address>.cnf file enable NAT:
# NAT/Firewall Traversal
nat_enable: “1”                ; <—- default is “0”. Have no idea why this works but…
nat_address: “”
voip_control_port: “5061”
start_media_port: “19000”
end_media_port:  “20000”
nat_received_processing: “0”

I chanced on this when poring over the SIP logs. I noticed that the requests from the phone was coming from random ports. The SIP read said…
<— SIP read from 192.168.2.130:50218 —>

… but the Contact: SIp field said…
Contact: <sip:6112@192.168.2.130:5060;transport=udp>

However, * responded…
<— Transmitting (NAT) to 192.168.2.130:50218 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.130:5060;branch=z9hG4bK4fba995f;received=192.168.2.130

Further down… i saw….
[Dec  8 17:32:01] VERBOSE[2565] logger.c: Sending to 192.168.2.130 : 50218 (NAT)    ; <—!!
[Dec  8 17:32:01] VERBOSE[2565] logger.c: Using INVITE request as basis request – 0008a326-31d90005-30de60e7-48868c82@192.168.2.130
[Dec  8 17:32:01] VERBOSE[2565] logger.c: Found no matching peer or user for ‘192.168.2.130:50218’

wierd?? Dunno. So * was sending its reponses to the 50218 port in the above case, and obviously the phone wasn’t listening at that port! This meant that the 200 OK responses were not being received by the phone. So finally hit on this setting… didn’t make sense, but tried it coz that would be one setting that would affect source and target ports. It worked! I hope this is the last of my travails with the Cisco 7940 phone. All works fine… including the MWI! 🙂

After this the sip show peers, showed me….
6112/6112                  192.168.2.130    D   N      5061     Unmonitored
^^^^-!!! This came from the NAT/Firewall traversal setting.
Obviously the Phone firmware was using a random port (i.e., ignoring this setting) when used with a nat_enable: 0  setting.

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