I am using 4 port analog FXO module with DAHDI. Incoming and outgoing calls works normally. But when softphone receives incoming call at softphone i can’t see the incoming number. Instead of incoming number there shows word “asterisk”. In with *.conf file i must change for determining incoming numbers.
I have been doing test, connecting 1 analog lines, to my Asterisk server trough fixed GSM with an FXS port. For this test i used 2 differents devices the first one shows the caller id correctly. But the other one shows the word asterisk as caller id. i will start playing with this issue, and i will let you know the solution in case i can find it.
Honestly i don’t know if they use FSK. And calling my PSTN won’t help. The tech support guy wont even know of what im talking about. So if they use FSK what do i need to do, And if the do not use FSK, what is the next step.
And i as said before I’m using 2 different FIXED GSM devices. One is Telular SX6T-GSM . The caller id works fine with this one. And the other one is a huawei b560. This last one, is the one that do not show the caller id. So it could be a misconfiguration or incompatibility with this last one device . because im using the same pstn line with both devices
You have to set dahdi.conf for the system they actually use. In extreme cases, that may mean writing new code for Asterisk, although I imagine most countries use one of the variations that Asterisk supports. If you really can’t discover the system they use, you will need to try all combinations of modulation scheme and start of CLID indication.
If none of them work, you will definitely need the technical specification if someone is going to add support to Asterisk.
Thanks David. But when you say dahdi.conf, you mean chan_dahdi.conf. According to this documentThere’s not any file called dahdi.conf in the dahdi configuration files.
you need to change the below options in chan_dahdi.conf usecallerid: Sets whether to use caller ID, “yes” or “no” are the only available options cidsignalling: Determine type of caller ID signaling in use. The Caller ID signaling types supported by Asterisk are:
bell: bell202 as used in US (default)
v23: v23 as used in the UK
v23_jp: v23 as used in Japan
dtmf: DTMF as used in Denmark, Sweden and Netherlands
smdi: Use SMDI for caller ID. Requires SMDI to be enabled cidstart: Determine signals the start of caller ID. The options supported by Asterisk are:
ring: A ring signals the start (default)
polarity: Polarity reversal signals the start
polarity_IN: Polarity reversal signals the start, DTMF dialtone detection in India
dtmf: DTMF Caller ID spill begins only with DTMF, at various times before the ring. This causes Asterisk to constantly listen for DTMF CallerID signals on the specified channels
Thanks striker24x7 for summarizing the information. I noticed that that i need to play with different combination, as David suggested before. I was using this links as reference voip-info.org/wiki/view/chan_dahdi.conf