search for: phone3

...on on '6a7f127b0d47ebd168678c665f4d2365@192.168.0.17' of Request 102: Match Found *CLI> Sep 13 22:18:10 DEBUG[13167]: chan_sip.c:6350 check_user_full: Setting NAT on RTP to 0 Sep 13 22:18:10 DEBUG[13167]: chan_sip.c:9413 handle_request_invite: Checking SI P call limits for device Phone3 Sep 13 22:18:10 DEBUG[13167]: chan_sip.c:5497 build_route: build_route: Contact hop: <sip:Phone3@192.168.0.18:5061> -- Executing VoiceMailMain("SIP/Phone3-9d74", "") in new stack Sep 13 22:18:10 DEBUG[13167]: channel.c:1388 ast_settimeout: Scheduling timer at 160...

...ow = all ; disallow = [101] device=SEP00B06409E748 model=12SP ; Valid models: 12SP, 30VIP, 7910, 7920 (so far) version=P002L2J2 host=172.20.1.1 context=telsip line => 101 callwaiting=no callerid="Juanjo",<101>; extensions.conf exten => 3,1,DBget(temp=clid/SIP/Phone3) exten => 3,2,SetCIDName(${temp}) exten => 3,3,DBdel(clid/SIP/Phone3) exten => 101,1,Dial(Skinny/101@101) When I make a call from Phone3(SIP) to the Skinny/101 the following messages appear in the Asterisk CLI: -- Executing Dial("SIP/Phone3-f631", "Skinny/101@101"...

...I have them configured correctly on asterisk to use sip channels, but when I call from one phone to the other I don't any voice communication between the phones. According to the phones I'm connected, but according to asterisk, I get the following message: -- Executing Dial("SIP/phone3-2c9f", "SIP/phone4") in new stack -- Called phone4 -- SIP/phone4-aaf5 is ringing -- SIP/phone4-aaf5 answered SIP/phone3-2c9f -- Attempting native bridge of SIP/phone3-2c9f and SIP/phone4-aaf5 == Spawn extension (sip, 2, 1) exited non-zero on 'SIP/phone3-2c9f'...

...1.3 localnet=192.168.10.0 localmask=255.255.255.0 [Phone1] type=friend host=dynamic nat=yes qualify=yes context=sip callerid="Phone1" <1> disallow=all allow=gsm [Phone2] type=friend host=dynamic qualify=yes context=sip callerid="Phone2" <2> disallow=all allow=gsm [Phone3] type=friend host=dynamic nat=yes qualify=yes context=sip callerid="Phone3" <3> disallow=all allow=gsm [Phone4] type=friend host=dynamic nat=yes qualify=yes context=sip callerid="Phone4" <4> disallow=all allow=gsm I use the following extensions for asterisk (extens...

...the log is the following - which displayed '601' on my phone. The caller was +886288097680 - am I getting the wrong ClID because of my end or the caller end? "","601","3","default","601","IAX2/guest@61.220.121.18:4569/1","SIP/phone3-99fb","Dial","SIP/phone3|30|tr","2005-01-03 16:53:33","2005-01-03 16:53:33","2005-01-03 17:02:00",507,507,"ANSWERED","DOCUMENTATION" Anyone care to call me? -- . . ___. .__ Posix Systems - Sth Africa. e.164 V...

I'm new in hire so Hello to everyone! I'm beginner user of Asterisk CVS-HEAD-10/01/04-14:31:34 . I just installed it on my Mandrake Linux 10.0 kernel 2.6.3-4mdk with sample configuration (used make samples). I would like to make phone connections between X-Lite (SIP) installed on computers in LAN. How to make this? I was reading manual, and tried to make changes in sip.conf but this all

Hi can i set up a group of SIP users and forward a call to it? I am looking for a group, not for a queue. I won't listen any musinc on hold, and i won't that someone has to pay if nobody of the user's in the group accept the call. Can i do that? Thanks to all

...s playing continuous on sequential extensions after a timeout, it is restarted for every extension in 1.2.14: ;music starts exten => 902,1,Dial(SIP/phone1@proxy.com|5|m(mymusic)) ;music starts again exten => 902,n,Dial(SIP/phone2@proxy.com|5|m(mymusic)) ;and again exten => 902,n,Dial(SIP/phone3@proxy.com|5|m(mymusic)) In the changelog this is not mentioned, also the bugs related to changes in musiconhold.c don't seem to have anything to do with my issue... Is there a setting i didn't see which changes this behaviour? It is quite annoying for the caller and unprofessional if he he...

I have an X100P used, at present, largely for outgoing calls. It shares the single incoming POTS line with a number of analog phones. Is it possible to talk the X100P (Zap/1) to answer a ringing call only if I ask it to? I'd like to use only the SIP phone in my office, but let the analog phones continue to work in the rest of the house (until I can afford FXS cards anyway..) I can force

...# extensions that are "not busy" ... sip.conf ... [2400] ; Grandstream Phone context=intern type=friend insecure=yes host=dynamic permit=192.168.254.0/255.255.255.0 mailbox=2400 dtmfmode=inband canreinvite=no nat=no ... extensions.conf ... PHONE2=SIP/2400 PHONE3=SIP/2410 RECEPTION=${PHONE2},${PHONE3} ... exten => 2200,1,AGI(myagi.agi,${RECEPTION}) ... console (asterisk -vvvvvvvc) ... -- Executing AGI("Zap/1-1", "myagi.agi|SIP/2400,SIP/2410") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/myagi.agi arg1 = SIP/2...

...ber in AstDB, then call that number exten => s-CFIM,1,DBget(WHEE=CFIM/temp) exten => s-CFIM,2,Dial(Zap/g0/${WHEE},20,tm) exten => s-CFIM,102,Voicemail,u203 exten => s-CFIM,103,Voicemail,b203 ; The 0 signifies no call forwarding. exten => s-NoCFIM,1,Dial(${PHONE1}&${PHONE2}&${PHONE3},10,tr) exten => s-NoCFIM,2,Wait(1) exten => s-NoCFIM,3,DigitTimeout(3) exten => s-NoCFIM,4,ResponseTimeout(7) exten => s-NoCFIM,5,Background(/var/lib/asterisk/sounds/hello) ;exten => s-NoCFIM5,Background(/var/lib/asterisk/sounds/vacation) exten => s-NoCFIM,6,Wait(1) exten => s...

...lls port=5060 bindaddr=0.0.0.0 srvlookup=yes [k-phone1] type=friend host=dynamic dtmfmode=rfc2833 secret=XXXXXXXX callerid="k-phone1" <601> context=internal [phone2] type=friend host=dynamic dtmfmode=inband secret=XXXXXXXX callerid="phone2" <602> context=internal [phone3] type=friend host=dynamic dtmfmode=inband secret=XXXXXXXX callerid="phone3" <603> context=internal Can anyone enlighten me? -- Pau

...username=user0 secret=secret1 auth=md5 host=192.168.3.60 context=incoming trunk=yes qualify=3000 disallow=all allow=ilbc voip-kntl:~# more /etc/asterisk/extensions.conf [general] static=yes writeprotect=no [globals] ; Global Variables ; Internal SIP Phone Numbers PHONE1=SIP/1001 PHONE2=SIP/1002 PHONE3=SIP/1003 PHONE4=SIP/1004 ; Other Site Authentication SITE1=IAX2/user0:pass0@site0 ; MACRO SECTION [macro-callextension] exten => s,1,Dial(${ARG1}) exten => s,2,Hangup exten => s,102,Playtones(busy) exten => s,103,Wait,30 exten => s,104,Hangup [internal-site1] ; Internal Extension...

...tandard Bell South phone lines. If I configure as so [channels] context=pstn group = 1 signalling = fxs_ks callprogress = yes channel => 4,3 Then any call routed from asterisk to the outside line will ring, and can be picked up, but * never detects pickup.. -- Executing Macro("SIP/phone3-e601", "dialout|93053484021") in new stack -- Executing Dial("SIP/phone3-e601", "ZAP/4/93053484021") in new stack -- Called 4/93053484021 -- Zap/4-1 is ringing -- Zap/4-1 is ringing I answer phone here.. * doesn't realize it... Voice is flowi...

...dial plan? I currently have 4 incoming lines going into a TDM400 with the group set to g0. Could it be that the way I've set this up, if any of the phones are busy, it goes immediately to VM? exten => s,1,Answer() exten => s,2,Wait(1) exten => s,3,Dial(${PHONE1}&${PHONE2}&${PHONE3},50,r) exten => s,4,Wait(1) exten => s,5,Voicemail,u203 exten => s,6,Voicemail,b203 exten => s,7,Hangup() Regards, Min

Okay, so like this? PHONE1+AD0-SIP/2000 PHONE2+AD0-SIP/3000 PHONE3+AD0-SIP/4000 ALL+AD0AJAB7-PHONE1+AH0AJgAkAHs-PHONE2+AH0AJgAkAHs-PHONE3+AH0- Then you would have Exten +AD0APg- s,1,Dial(+ACQAew-ALL+AH0-,20) Is that right? I have read about the Macros but don't understand their use. Could someone provide an example? Sorry about the newby questions......

Whilst on a call, I'm getting the following... -- Started music on hold, class 'default', on SIP/phone3-a7d5 -- Playing 'pbx-transfer' (language 'en') -- Unable to find extension '#' in context 'default' -- Playing 'pbx-invalid' (language 'en') ie - without anyone pushing keys - I hear the music on Hold - as does the calling party. Are we...

...mailto:markster@digium.com>> ========================================================================= [ Booting............ -- SIP Seeding 'phone1' at phone1@192.168.0.158:5060 for 3600 -- SIP Seeding 'phone2' at phone2@192.168.0.176:5060 for 3600 -- SIP Seeding 'phone3' at phone3@192.168.0.167:5060 for 3600 .......Jun 29 11:56:33 WARNING[-1084538752]: chan_skinny.c:2541 reload_config: Unable to get our IP address, Skinny disabled ..Jun 29 11:56:33 WARNING[-1084538752]: chan_zap.c:704 zt_open: Unable to specify channel 1: No such device or address Jun 29 11:56...

...r of preference ;allow=ilbc ..... [Phone1] type=friend host=dynamic defaultip=192.168.3.103 dtmfmode=rfc2833 context=from-sip callerid=" Win box " <1> [Phone2] type=friend host=dynamic defaultip=192.168.3.119 dtmfmode=rfc2833 context=from-sip callerid=" Deepak" <2> [Phone3] type=friend host=dynamic defaultip=192.168.3.106 dtmfmode=rfc2833 context=from-sip callerid=" Ravi " <3> [extensions.conf] [from-sip] exten=>1,1,Dial(SIP/Phone1,20,tr) exten=>2,1,Dial(SIP/Phone2,20,tr) exten=>3,1,Dial(SIP/Phone3,20,tr) ----------------------------------...

...number: 415 541 XXXX. If it does not work we can also use his/her mobile number. We need to manage more than 180 agents nationwide so I would like to use a table or data base to translate a large number of agent's telephones. The table looks like this: EXT PHONE1 PHONE2 PHONE3 3021 4155XXXX 415YYYYY 510XXXXX 2130 415ZZZZZ 510LLLLL 3060 510XXXXX XXXXXXX nnnn . . xxxx XXXXXXX XXXXXXX Thanks in advance for your help. Ypek _________________________________________________________________ Sadece sohbet ile yetinmeyin - eglneceye de doymak...