Missing sounds in Asterisk

Written late in the afternoon in English • Tags: , , ,

The latest odd fix for Asterisk after an upgrade (11.13.0~dfsg-1~bpo70+1):

cd /usr/share/asterisk/sounds && ln -s en_US_f_Allison en

I don’t know when this broke. I found out because calling voicemail would fail (due to a missing password prompt sound file).

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As of Asterisk 1.8 a prefix is no longer valid for choosing the busy / unavailable announcement: VoiceMail(u${ARG1}) You need to pass it in the options instead: VoiceMail(${ARG1},u)

Properly overriding caller ID

Written at lunch time in English • Tags: ,

I don’t usually call myself so I hadn’t until recently made note of my own caller ID not being what it should. Looking at CDR logs the caller ID on one of my phones changed in September 2012 from “my” extension to the extension of the phone device. (more…)

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Linksys SPA942 phones reboot on DHCP lease renewal even if the lease information does not change (apart from the obvious timestamps). Increased lease time to 1 week for all phones.

Time to upgrade Asterisk

Written late in the evening in English • Tags: , ,

SIP appears to be broken in the Asterisk 1.6 packages released to address DSA-2550. I resorted to a hasty upgrade to Asterisk 1.8 from backports. (more…)

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The Linksys SPA3102 has become difficult to find, so I’ve replaced one failed unit with a SPA2102. Seems to work just as well and its configuration is practically identical.

Bring back audio after Asterisk 1.6 upgrade

Written in the mid-afternoon in English • Tags: , ,

I upgraded to Asterisk 1.6 some time ago, but didn’t think anything was wrong until recently. Calls coming in from Callcentric didn’t work: I received no audio. Everything had been working fine with Asterisk 1.4. I don’t get many calls, so initially I dismissed this as a temporary problem. Calls from my other four carriers kept working fine.

After some research, I noticed the following settings suggested by Callcentric:

session-timers=refuse
session-expires=180
session-minse=90
session-refresher=uas

I’ve placed this in the [general] section of sip.conf, because calls from Callcentric arrive from multiple servers and the way Asterisk handles SRV records, only one of the servers ends up mapping into the per-carrier context at any given time. It doesn’t seem to have an adverse effect on calls from other carriers. (It is just turning off functionality new to 1.6, and setting some sensible defaults.)

Ekiga on the intranet

Written in the mid-afternoon in English • Tags: ,

Ekiga has a problem talking from behind a firewall, or maybe the problem is that both the firewall and Ekiga are trying to be too clever. I’m running siproxd on the firewall to transparently handle SIP connections. All the hardware phones and ATAs as well as X-Lite work well, but Ekiga fails to register.

The fix is controversial: I’m disabling STUN.

gconftool-2 -s /apps/ekiga/general/nat/stun_server --type=string

However, this means Ekiga won’t work from less “intelligent” networks anymore. Since I’m considering Ekiga for my laptop, this might be a problem. Maybe I should try X-Lite under Wine.

Nokia E65 on Asterisk

Written late in the afternoon in English • Tags:

I reconfigured my Nokia E65 mobile phone to connect to my Asterisk PBX, so I can make and receive SIP calls instead of cellular calls when I’m at home or at the office. I already had the corresponding WLANs configured on the phone for web browsing, so I didn’t have to do that part now (and thus those settings are not documented here). (more…)

Tuning the phones

Written early in the evening in English • Tags:

I spent some time with my parents about a month ago fine tuning the echo cancellation on our SIP phones. Today I got a chance to refresh my memory while connecting the phones of a friend to a CallWeaver PBX I had setup for him. Hopefully I have now collected in one place all the settings that were changed from the factory defaults. If you have a Linksys SPA3102 phone adapter or a Linksys SPA942 phone, you may find some useful hints in here. (more…)