Actually, the code below works perfectly to fix the transfer disconnect problem. I was asking of other, better ways, aside from manually defining on all incoming calls a dummy CID.
To answer Steve's question, using a single TDM400 card for the incoming PSTN (it's one line, a remote office that most of their communication is done over IAX back to our main location). The three handsets are Grandstream GXP-2000 (let the flaming begin, we currently have about 40 GXP-2000's in production and yes, we've had strange issues, but they're working quite well now). Anyway, it's really not a huge deal, but I had work arounds. I'd prefer the 'usecallerid=no' type route instead of making a fix in the dialplan, that's all I was looking for. Ken -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Friday, April 25, 2008 7:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No CallerID Transfer Problem Try removing the quotes from the Caller*ID info. Steve Davies wrote: > 2008/4/24 Ken Williams <[EMAIL PROTECTED]>: >> Came upon a problem today that I thought I'd see if it's by design, >> if I'm missing an option somewhere, or if my fix is the way to fix it. >> >> We setup a remote location with a server, same as we've done with >> others, but for some reason when they would transfer an outside call >> anywhere it would pause for a few seconds and hang up the line. >> >> Well, after spending most of the day on it, it turns out it's because >> they don't have callerID on the PSTN lines coming in through zaptel. >> My first thought was, set "usecallerid=no" and all would be well, but >> this didn't do any good. After playing a bit longer I just set the following: >> >> exten => 900,2,set(CALLERID(num)="606-555-1212") >> exten => 900,3,set(CALLERID(name)="Outside Call") >> exten => >> 900,4,Dial(${DIALEXTENSIONS},${RINGTIMER},${DIAL_OPTIONS}) >> >> Now all works well. >> >> So is there another option somewhere to keep asterisk from killing a >> transfer without callerid? This happened on both 1.4.17 & 1.4.18.1. >> >> Thanks, >> Ken > > Can I guess that they are using snom phones with firmware 7.1.30? I > encountered exactly that bug here, but only if I enabled "sendrpid" in > the sip.conf of the asterisk system. Downgrading to a more-stable > 6.5.x snom firmware, or disabling "sendrpid" for all of the snom > devices fixed this in our case. (Roll on the next snom firmware > release!) > > If not, then can I suggest that you provide more detail of equipment > involved - PCI cards, handsets etc etc? > > Hope that helps, > Steve > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
