search for: phone1

...wing error : * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module loaded, can't setup SRTP session. This is a working internal call : > == Using SIP RTP CoS mark 5 > -- Executing [301 at local:1] Dial("SIP/dbucher-00000000", > "SIP/phone1") in new stack > == Using SIP RTP CoS mark 5 > -- Called phone1 > -- SIP/phone1-00000001 is ringing > -- SIP/phone1-00000001 is ringing > -- SIP/phone1-00000001 is ringing > -- SIP/phone1-00000001 is ringing > -- SIP/phone1-00000001 is ringing &gt...

...ar sip registrar port: 5060 # 5060 is set by default. sip digit time out: 6 time server disabled: 0 # Time server disabled. time server1: 192.168.0.90 # Enable time server and enter at <mac>.cfg - this is the correct mac address in all uppercase sip line1 auth name: phone1 sip line1 password: 1234 sip line1 registrar ip: 192.168.0.94 sip line1 user name: phone1 sip line1 display name: "myname" sip line1 screen name: "myname" sip.conf [general] port = 5060 bindaddr = 0.0.0.0 context=tutorial [phone1] type=friend username=phone1 secret=1234 host=d...

...there is the following error : * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module loaded, can't setup SRTP session. This is a working internal call : == Using SIP RTP CoS mark 5 -- Executing [301 at local:1] Dial("SIP/dbucher-00000000", "SIP/phone1") in new stack == Using SIP RTP CoS mark 5 -- Called phone1 -- SIP/phone1-00000001 is ringing -- SIP/phone1-00000001 is ringing -- SIP/phone1-00000001 is ringing -- SIP/phone1-00000001 is ringing -- SIP/phone1-00000001 is ringing -- SIP/phone1-00000001 answered SI...

...ply for my 7960 and am having problems getting it working. What should be in sip.conf and the SIP(macaddr).cnf file? This is what I have in SIP0002FD3BA8F7.cnf # SIP Configuration Generic File # Line 1 appearance line1_name: Asterisk Test # Line 1 Registration Authentication line1_authname: "phone1" # Line 1 Registration Password line1_password: "phone1" And sip.conf contains: [phone1] type=friend secret=phone1 host=dynamic defaultip=192.168.1.28 dtmfmode=inband I am trying to call between the 7960 and a Grandstream phone. It would work from Cisco -> Grandstream, but not v...

...ng nothing. I looked at sip debug (below) but am new to Asterisk and don't really know what I am looking for. Asterisk works fine with XLITE so I know my installation is ok. Sip read: INVITE sip:90800500005@192.168.0.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-Ifn-9746 From: phone1 <sip:phone1@192.168.0.250:5060>;tag=eSJ-4736 To: <sip:90800500005@192.168.0.250> Call-ID: diY-24872@192.168.1.1 CSeq: 1 INVITE Contact: <sip:phone1@192.168.1.1> Max-Forwards: 70 User-Agent: DrayTek UA-1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Type: application/sdp Conte...

...tmp/smstest ${EXTEN} ${CALLERIDNUM} ${CALLERIDNAME}) exten = _X.,3,Hangup When trying to send an SMS from the phone to *, I get the following: /var/log/asterisk/event_log: asterisk[15345]: No data, hanging up asterisk -grv with verbosity set to 7 shows this: -- Executing Goto("SIP/phone1-3037", "smsmorx|101|1") in new stack -- Goto (smsmorx,101,1) -- Executing SMS("SIP/phone1-3037", "101|sa") in new stack -- SMS TX 93 00 6D 00 00 00... -- Executing System("SIP/phone1-3037", "/tmp/smstest 101 101 Office") in new...

...ake a conversation. = ok When I call to a softphone with a Grandstream I can pich up the call with the softphone but the Grandstream keeps ringing like on the other site you didn't pick up the phone.(even if you do so) It's the same when I call between two Grandstream phone's. Call from phone1 to phone 2, I pick up phone2 and afther 3 seconds I get congestion tone from both phone's. Info from command *CLI> -- Executing Dial("SIP/phone2-a030a", "sip/phone1") in new stack -- Called phone1 -- SIP/phone1-663a is ringing -- SIP/phone1-663a answered SIP/phone2-a030a...

...sk doesn't support), i've set all my two snom 300 phones to support only g722 and asterisk declined the sip invitation. That is bad for me. Is it possible that asterisk asks the called phone not the codecs that asterisk supports but the codecs that the calling phone supports? What i want: phone1->asterisk: hello, i'm calling phone2, codecs possible: g722 asterisk->phone2: hello, you have a call from phone1, codecs possible: g722 phone2->asterisk: ok, let it be g722 ... chit chat ... But asterisk does this: phone1->asterisk: hello, i'm calling phone2, codecs possible:...

Hi, I've have followed through the help docs in trying to get an initial setup going with two phones and the asterisk server. Firstly, all I'm trying to do is get the two phones actually talking to one another VIA asterisk.. I've added this to sip.conf: [phone1] type=friend host=dynamic defaultip=192.168.1.106 ;username=blah ;secret=blah dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info mailbox=1000 ; Mailbox for message waiting indicator context=sip callerid="Me" <2124> [phone2] type=friend ;secret=blah host=dynamic defaultip=192....

Dear, When I start asterisk -vvvvvvgrc and I ask 'sip show peers', I don't get the ip adress in the 'Host" field. Name = phone1 and phone2 Host=unspecified mask 255.255.255.255 port = 0 status = unmonitored I can ping the two phone's and get a reply (also from the laptop) phone ip adres 192.168.10.12 and 192.168.10.13 (server 192.168.10.11and laptop 192.168.10.14) hardware config: server - phone1 - phone2 - laptop c...

...39;y' and 'z' are optional. The following special variables are optional for limit calls: (pasted from app_dial.c) w: Allow the called user to start recording after pressing *1 or what defined in features.conf (Asterisk > v1.0.x) for example I've tired: exten => 21,1,Dial(${phone1},20,r,w) exten => 21,1,Dial(${phone1},20,r,L(5[:4][:1])) but none is working. I've *-1.0.5 but I can not find app_dial.c nor features.conf contains any define recording options. -- #Joseph

Hi, Actually on my setup all outgoing calls are going trhu a SIP unique account A have a second SIP account with another operator and I would like my setup to use alternatively each of the two accoutns Call 1=> Dial SIP/phone1 Call 2=> Dial SIP/phone2 Call 3=> Dial SIP/phone1 <...> If you have an sample please let me know

Hello all, Here my architecture : Proxy1-asterisk1-proxy2-phone1 If a call arrived from proxy1 to phone1 AND phone1 always forward to proxy, asterisk1 say: -- Now forwarding SIP/phone1-0000001d to 'Local/969990349 at proxy2' (thanks to SIP/proxy2-0000001e) Why it use Local ? I just need to use as a normal call, not a local Thank you Fra...

I've just installed asterisk on a Debian Linux (apt-get it) And i have placed two sip phones in sip.conf and i'm testing parking with them I have phone1-SIP/1000 and phone2-SIP/1007 The following happens if i park from calling party and everything is OK 1. Pickup Phone2 and call to Phone1 2. Talk 3. Phone2 dials #700 and parks the call (it is placed in 701) 4. Phone2 is hangup 5. Pickup Phone2 and dial 701 6. The Phone2 is connected back to phone1...

Dear All, I installed an Asterisk on a linux PC, and X-Lite on two Windows PCs, all in a LAN. But, when I make phone call from one X-Lite to another, I always get Call Failed: 404 not found. Here is my sip.conf: [Phone1] type=friend host=dynamic ;defaultip=192.168.1.103 dtmfmode=rfc2833 context=SIP callerid = "Me" <2124> [Phone2] type=friend...

I don't know if this is possible, and I can't quite get my head around how to do it... If I am using DUNDi for redundancy in a cluster, when Phone1 makes a call to Phone2, both Asterisk A and B will be in the RTP stream: +---+ +---+ | A |-----| B | /+---+ +---+\ / \ Phone1 Phone2 Is there a way configure re-invites in this situation so that either Asterisk A or B drops out...

...in the config but am not getting anywhere This is what I'm getting from console (user/pass/dest # changed for obvious reasons): DEBUG[1133735216]: File chan_sip.c, Line 3841 (check_user): Setting NAT on RTP to 0 DEBUG[1133735216]: File chan_sip.c, Line 4891 (handle_request): Check for res for phone1 DEBUG[1133735216]: File chan_sip.c, Line 973 (find_user): Call from user 'phone1' is 1 out of 0 DEBUG[1133735216]: File chan_sip.c, Line 3307 (build_route): build_route: Contact hop: <sip:phone1@10.1.2.24:5060;line=1> -- Executing Dial("SIP/phone1-2c71", "IAX/user:s...

I am seeking to implement the following behavor: When a headset on phone1 is picked up, phone2 rings right away, without any need to dial numbers on phone1. Is this possible to implement? ScriptHead -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060120/0892441d/attachment.htm

...=> 40,1,Answer Exten => 40,2,SetMusicOnHold(default) Exten => 40,3,DigitTimeout,5 Exten => 40,4,ResponseTimeout,10 Exten => 40,5,Background(greeting) Exten => 1,1,Playback(secr) ; if you press <91>1<92> playback message <93>secr<94> Exten => 1,2,Dial(SIP/Phone1/20) Exten => 2,1,Playback(studentservice) Exten => 2,2,Dial(SIP/Phone1/20) Exten => 3,1,Playback(it) Exten => 3,2,Dial(SIP/Phone1/20) Exten => 4,1,Playback(operator) Exten => 4,2,Dial(SIP/Phone1/20) Inside asterisk debug i see what the forwarding of the call working : log of A...

...o there, I appreciate any help about this problem that I can't figure out... I need to record all my calls: this is pretty easy using Monitor() before the Dial(). eg: exten => 425,n,Monitor(wav49,/var/spool/asterisk/monitor/425/${EPOCH}_${CALLERID(num)}_in,mb) exten => 425,n,Dial(${PHONE1},10) Now, I want to create a call group: I mean, I want a number (eg 800) that makes 3,4...10 phones ringing together. I found 2 modes to do this: 1) exten => 800,n,Dial(${PHONE1}&${PHONE2}&${...},15) 2) with Asterisk 1.6: exten => 800,n,Set(DIALGROUP(test,add)=${PHONE1}) exten =&g...