similar to: Rewrite calling number of incoming call

Hi all! I am trying to figure out how DEVICE_STATE is working, no luck so far. sip.conf [0317998975] type=friend regexten=0317998975 secret=???? username=0317998975 callerid="Magnus Benngard" mailbox=0317998975 at inputinterior.se host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes disallow=all allow=alaw extensions.conf exten => 0317998975,hint,SIP/0317998975 exten =>

Hi All, I'm having some troubles with using call files. I'm trying to establish the following: - want to use call files to connect two (outside) extensions - want to use the outbound routes set in FreePBX - want to set the outgoing callerid for both calls - want to set a custom CDR field in MySQL ( field name 'azonosito' ) Asterisk is version 1.8.2.3 with freepbx 2.8.1. What

Hi, I tested the capi_chan with latest cvs of * and I have problems with Early-B3. The following dialstring works for me (without Early B3): exten => _0X.,1,Dial(CAPI/@22715291:${EXTEN:1}|30) But if I add the 'b' for using Early-B3 exten => _0X.,1,Dial(CAPI/@22715291:b${EXTEN:1}|30) nothing changes (no dialtone). If in this example the called party discards the call, there is no

Hi and thank you. I am a beginer trying to install my first TDM04B. I am able to receive call with the card using: [incoming] exten => s,1,Dial(SIP/robgol,20,tr) on my extensions but, with [outgoing] exten => _0X.,1,Zap/1/${EXTEN} I cant send them out. I am getting the following error: Mar 26 23:37:07 WARNING[5189]: pbx.c:1291 pbx_extension_helper: No application

Hi! Trying to figure out what I am doing wrong... 1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256) 1 Cell phone 00733025975 attached through H323. extensions.conf exten => 975,1,Goto(975-${DEVICE_STATE(SIP/0317998975)},1) exten => 975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs) exten => 975-INUSE,2,Hangup() exten =>

Gentlemen, Did tryout "CONNECTEDLINE" function, was exactly what I have been looking for. But there are at least one thing I cant figure out. Did a very simple and "stupid" extension 0317998955 and ran a test. My phone (0317998975) dials 955, the display on my phone changes from "955" to "Connected Line 955" when my call is answered, shouldn't the

Hi, Urgently response would be wonderful, system is a Fedora Core 2. I have a Ericsson BP250 connected to 1 port on the TE405P and another connected to a local telco ISDN30. I have been running CVS-HEAD from about a 2 months ago and upgraded it again just in cause it was a version issue (didn't fix it) but this is what I am getting. When a person calls out from an extension on the BP250 to

Hi everyone, having a issue with asterisk and my new Voip providers service. Iv set up many asterisk systems before but never seen this and have tried to fix this with no luck.. I have used this exact same sort of setup for 5 other providers and never had this issue, If i replace the trunk login details with my works voip account and set it to IAX then it works perfect, Just not the new

Hello Gentleman, I'm new to asterisk, this is my first instalation, first post... so I'd like to apologize if this question has already been made. I googled but I couldn't find nothing similar. Here's the thing. I'm migrating from ATA to Asterisk one of my client's office and I have a very simple setup. A Linux PC running Debian Lenny + Asterisk 1.4.21. It's a

Hi! We have alot of users who are having 2 phones, 1 fixed and 1 DECT. I am looking for a way to log them into a queue and let both phone rings. Let me try to explain: 0317998975 is a fixed phone, 0317998985 is a DECT. 0317998989 is a queue. queue add member SIP/0317998975 to 0317998989 works ofc. sip*CLI> queue show 0317998989 0317998989 has 0 calls (max unlimited) in

Gentlemen, Forgive me if I am posting at the wrong place! I was going to test the "new" chan_ooh323 driver so I did install: debian: Linux sip2 2.6.26-2-686 #1 SMP dahdi-linux-complete-2.2.0.2+2.2.0 Asterisk SVN-trunk-r231692 Did enable chan_ooh323, everything compiled without any problems. Hardware setup: Phone (975) - Avaya CM - H.323 - Asterisk - X-Lite (0317998975) X-Lite can

How to make outgoing calls from DID and what is theway to get incoming calls from DID. -- ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110526/3f19091d/attachment.htm>

Hi! Have a weired problem with Avaya 9650 phones: extensions.conf exten => 0317998975,hint,SIP/0317998975 exten => 0317998975,1,Goto(0317998975-${DEVICE_STATE(SIP/0317998975)},1) exten => 0317998975,2,Hangup() exten => 0317998975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs) exten => 0317998975-INUSE,2,Hangup() exten => 0317998975-NOANSWER,1,VoiceMail(0317998975 at

HI, I'm using the usual Set(Callerid(num) function to change the incoming from skype callerid but it's not working. Asterisk 1.4.31 and last release of skype channels This is the dialplan exten => _0X.,1,NoOP(${CALLERID(num)} - ${CALLERID(name)}) exten => _0X.,n,Set(STRINGA="Skype") exten => _0X.,n,NoOP(${STRINGA}) exten => _0X.,n,Set(CALLERID(num) = ${STRINGA})

I have my peers registered to SER.asterisk seems to be sending mwi for the peers seen in the sip show peers CLI command. i have my ser server registered with asterisk as a type=friend and all clients register to ser.how do i get mwi to work for these clients registered to SER. Thank you, -AA

Hi, Running Asterisk CVS-Head latest on a Dual P3 800 1Gb Ram. For some odd reason now that I have the asterisk box almost to the stage I want it, I hit a problem. I have a te405p in the system, Zap/g1 is connected to the telco as an ISDN 30, Zap/g4 is connected by ISDN Primary Rate to an Ericcson BP250 phone system. When calls come in on g1 they go straight through instantaneously to the

For make outgoing call, i setup 0. However 0 is write in the cdr dst field. Is there a way to remove it when asterisk send it to cdr_mysql ? exten => _0X.,1,Dial,SIP/${EXTEN:1}@mygateway I just want have in cdr dst = ${EXTEN:1} This don't work : exten => _0X.,1,SetVar(EXTEN=${EXTEN:1}) exten => _0X.,2,Dial,SIP/${EXTEN}@mygateway Use another variable still record ${EXTEN} --

For make outgoing call, i setup 0. However 0 is write in the cdr dst field. Is there a way to remove it when asterisk send it to cdr_mysql ? exten => _0X.,1,Dial,SIP/${EXTEN:1}@mygateway I just want have in cdr dst = ${EXTEN:1} This don't work : exten => _0X.,1,SetVar(EXTEN=${EXTEN:1}) exten => _0X.,2,Dial,SIP/${EXTEN}@mygateway Use another variable still record ${EXTEN} --

Hello all, With some help from people in #asterisk on freenode, I've managed to get incoming SIP calls working. Outgoing calls however are however a different matter. My whole working (incoming calls only) SIPgate configuration can be found here. [1] When I uncommon what's in there, nothing works. There doesn't appear to be any useful error being logged , even when debug is

Vinicius Fontes wrote: > Hey guys, I don't know if this is the right place to ask this. I was > thinking about reporting a bug, but maybe it's better to sort out if > this is really a bug or just me being lame. > > I want to record *every* call in my Asterisk box, so I use the > MixMonitor() application like this is my extensions.conf: > > exten =>