Brian McCrary
2004-Nov-23 06:34 UTC
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 4, Issue 300
Andrew Thompson wrote:> You should be able to set the inbound callerid from the switch/gateway > to a specific unknown in sip.conf file with just a callerid= line. > > The place I looked on the wiki didn't show a specific description for > the callerid= line, but that's what I thought I read for it somewhere. > > http://www.voip-info.org/wiki-Asterisk+config+sip.conf (currently hosed) > http://64.233.179.104/search?q=cache:IIOmLeG89KwJ:www.voip-info.org/wiki-Asterisk%2Bconfig%2Bsip.conf+site:voip-info.org+sip.conf > (google cache)Thanks for the idea!! I tried putting the callerid=Unknown line (also tried it with an actual phone number) in the sip.conf file where my gateway config is at, but it didn't seem to help any. If anyone has any ideas I'd appreciate it, I might can modify the gateway by putting in a translation rule so if the calling number is blank it will put something in there, but I would think Asterisk would handle that. I have used VOCAL for awhile and it works fine with this, but I know it's relaying SIP directly to the phone where as Asterisk acts as a "buffer" between them. I think otherwise Asterisk is an outstanding platform that is very feature rich, and I'm happy to say with help of others on the list, I have it working quite well under Solaris in a telecom environment. Thanks, Brian
Following up on an issue I had posted about earlier, I was able to find a solution. I had the gateway defined as [as5400] in the config file. Since the name did not match it's IP address, when Asterisk started looking for Caller ID info, it couldn't find any. I made an entry into the sip.conf file that has the IP address of the gateway in it, and it is now returning that if the number is unknown. I only put 2 lines in the section, so it looks like: [xx.xx.xx.xx] ; IP address of gateway type=friend callerid=Unknown <000-000-0000> Just thought I'd share with the list in case someone else needs to know this in the future. Brian
I am just learing some Linux and have been able to setup Asterisk samples and channels fxo card on ch.1 and fxs on ch 4. I have an Internet Polycom phone to use to test to/from internet and 1 analouge phone connected to port 4 of Digium TDM-400 with appropriate cards installed to dial out on. I wish to dial to the outside via PTSN line. I am lost on the instructions. Can anyone help with Extensions.conf and sap.conf. 3 extensions are needed. Thanks for help. Leo