If you want to run a business in China

...then you will need a Chinese phone number. I.e. a phone number with the country code +86. Your customers will use this number to reach your company, and you will use this number for outgoing calls to them, too.

There are many SIP providers that offer Chinese phone numbers, but not all of them are good. Here is why.

The phone system in China has an important quirk: it mangles Caller ID numbers on incoming international calls. This is not VoIP specific, and applies even to simple mobile-to-mobile calls. E.g., my mobile phone number in Russia starts with +7 953, and if I place a call to almost any other country, they will see that +7 953 XXX XXXX is calling. But, if I call a phone number in China, they will instead see something else, with no country code and no common suffix with my actual phone number.

The problem is that some SIP providers land calls to China (including calls from a Chinese number obtained from their pool) on gateways that are outside China. If you use such provider and call a Chinese customer, they will not recognize you, because the call will be treated as international (even though it is intended to be between two Chinese phone numbers), and your caller ID will be mangled.

As far as I know, there is no way to tell if a SIP provider is affected by this problem, without trying their service or calling their support.

Recording SIP conversations

My chief prefers to use SIP (voice conversations) instead of Jabber (text messaging). The problem is that he sometimes overloads me with a lot of information, and later I forget 90%. This is not a problem with Jabber, as there are logs. However, I don't know any SIP client that supports recording of conversations out of the box.

Here is my solution. You need alsa-lib 1.0.17 or later (i.e., if you are using Debian Lenny, due to the freeze, you'll have to fetch libasound2 and libasound2-dev from experimental). Create the subdirectory named "voice" in your home directory. Add these lines to $HOME/.asoundrc:

pcm.voice {
        type asym
        playback.pcm {
                type file
                file "/home/user/voice/p"
                slave.pcm "default"
                format "wav"
                truncate 0
        }
        capture.pcm {
                type file
                file "/home/user/voice/c"
                slave.pcm "default"
                format "wav"
                truncate 0
        }
}

Then, configure your SIP client (I use twinkle) as follows. ALSA device for rings: "default", ALSA speaker: "voice", ALSA microphone: "voice", Play back ring tone when the network doesn't: disable. As the result, ALSA library will write the following files into the voice directory: p, c, p.0001, c.0001, p.0002, c.0002, and so on (the number is incremented for each new call). All these files are in wav format, "p" files correspond to the remote side, and "c" files contain what you spoke into the microphone. It appears that it is impossible to record both parties of the conversation into one file using only built-in features that come with ALSA.

To play back the conversation, do:

aplay p.0002 & aplay c.0002

This will cause two aplay processes to be started in parallel, with their outputs mixed. Or, you can make one mixed wav file with sox:

sox -m c.0002 p.0002 -f wav talk0002.wav