similar to: When callerid changes its value ?

Hello all, I'm having a problem with getting incoming callerid to a lan-connected phone. The Asterisk server is connected to the Internet, and a Grandstream BT101 phone on a lan interface: INTERNET ----(eth0) Asterisk (eth1) ---- Grandstream (192.168.1.51) The phone registers with the Asterisk server as ext 20. I can initiate and receive calls from the Grandstream phone fine. The

As far as I can remember I only opened sip and tftp ports for the phone. For some reason (didn't look into it too much) the call stays with sip and doesn't use RTP. The problem you describe (the call doesn't even ring on the other side) is something I had and was solved by upgrading the firmware. Checkpoint's tracker explicitly said what connection attempts were blocked and why.

dear all, can anyone assist on how to enable callerID to appear on grandstream bt-100 sip phone. [1003] type=friend context=default username=1003 secret=**** ;fromuser=1003 ;callerid=John Doe <1234> host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 incominglimit=1 ;mailbox=1234@default ; mailbox 1234 in voicemail context "default" disallow=all allow=ulaw allow=alaw

Check your FW-1 tracker and see if any sip packets are dropped during call initiation. I had this problem and it went away when I upgraded the BT's firmware to the latest (16). Beware, though, that people on the list claim that this firmware breaks functionality of the message button and autoanswer. I haven't checked this yet, cause I can't afford to go back a version. I prefer a

Hi to all, can you suggest to me the best way to avoid problems in the CDRs for anonymous sip calls? I have some peoples that set Send Anonymous : Yes in their Grandstream phones and i don't receive the username as phone number that i use to make billing. It is empty. The only place where there is the phone number is in the peer name where it write the name of the peer that in this case is

hi, I have a problem with passing caller id information from telco (isdn) to sip client (grandstream). i see callerid in asterisk verbose console but on grandstream (sip) phone is just internal (own-gs) 101 number. Isdn line is connected with hfc card and p2p , asterisk is latest CVS in extensions.conf i have: exten => 2442242,1,Dial,SIP/101,r|T and in console is this: -- Executing

Hi all, I want to pass the incoming SIP callerid in Dial application: Asterisk 1.2.13 sip.conf: register => user:pass@provider/ext extensions.conf: exten => ext,1,Dial(SIP/phone1&SIP/phone2) on phone's display I see the 'ext' number, not the incoming SIP callerid as can be seen on incoming calls when I register the phone directly to provider. I tried to add

Hi, I'd like to organize my Asterisk to properly handle incoming SIP/IAX/H323 callerids so they can be called back if needed. I have three incoming contexts for sip, iax and h323 calls. To each incoming call I'd like to prepend certain number that will be catched with pattern matching on output calls. For instance for iax I have: [from-iax] exten => s,1,NoOp(IAX call from outside

I have some phones (and an ATA) that are shared between two users who each have separate voicemail but they are not behaving as desired nor expected. Incoming calls show up on the correct lines. Calls originating from the device are seen, at the terminating device, as coming from the account listed last in sip.conf, regardless of the line selected. This creates three main issues I would like

Heres a weird one... Call comes in on mISDN channel. Little bit of dialcode (in a macro) looks up the number in the astdb and puts an name to it. No real magic there, and it works well. Same macro also has parameter passed in to put a prefix on the name - this is set in the DDI handling and is dependent on the number called and allows phone users to see which number was called (company

Hi everyone, I have been dealing with a certain issue with a particular customer site for months now. The problem occurs when there is an error with caller id as shown in the following: WARNING[16036]: chan_zap.c:6309 ss_thread: CallerID returned with error on channel 'Zap/3-1' When this happens, it appears that the call still goes through as I can see the caller still navigating

Hi All, I have a server setup with an incomming PSTN line and a bunch of Grandstream BT100 phones. Is there a way for asterisk to foward an incomming callerID from the PSTN to the SIP phone that is setup as an extension? We have a Voice menu setup for incomming calls and I would like to recieve the caller ID of the calls we are recieving after the incomming caller reaches their final

Thanks to everyone for their help. I sorted out my CallerID problem - I had a stray "fromuser=101" command in my sip.conf which was overwriting any CallerID info. It was a process of elimination (on my part) helped by all the comments I had back. Regards, George

I've dealing with this issue for a while, and I'd really like to know if anybody has experienced the same pain before :-) I've a lot of Grandstream HandyTone 286, loaded with the latest firmware (1.0.8.16) from the GS website. In my sip.conf, this ATA's are configured as: [05] type=friend username=05 secret=XXXX callerid="User 05" host=dynamic nat=yes qualify=yes

I have the following: Call -> PSTN -> * -> GrandStream 101 1.0.3.81 The GS displays "ohn ro n2600" when the call is past to the GS. If I pass the call to a XTEN client, Caller ID shows up. Any ideas ??

Hi, Does anyone know anything about the callerid format that NTL uses here in Cambridge, UK. This is the former Cambridge Cable, who is sometimes different from the rest of NTL. They did say that only some equipment works with their switch. I hoped that they might use US-style CID, which would work with the X100P, but it doesn't seem to come through. I do notice that my DECT phone bought

I have a site running asterisk 1.2.8 with a hand full of polycoms and grandstream 2Kxp's. When a call is missed and you look at the missed call logs on either, its has the persons exten, not the incoming caller id. Any ideas? \\\|/// \\ ~ ~ // ( @ @ )--oOOo-(_)-oOOo? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying to send CallerID info to a MetaSwitch system over a PRI. The MetaSwitch gets the info exactly as it is sent by Asterisk, but I think it might be having trouble with the Hexadecimal b1 that is being sent just before the first character of the CallerID Name. Does anyone know what the significance is of the b1 being sent here? Or, is there a way to make Asterisk not send the b1

Hi, I have two local SIP extensions (both bt100). One is on remote location behind another nat (16), but everyithing seems to be setup correctly as it can register and is listed as OK(57ms). However I can only call in one direction between those two. Extensions are defined in same context: exten => 11,1,Macro(oneline,SIP/11) exten => 16,1,Macro(oneline,SIP/16) both using same macro

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