search for: abouttotry

...'s some other related issue. So I try to use the demo and do "dial 500". This should connect and play the demo's intro file. I've gotten this to work in the past. Instead I get the following warnings: > WARNING[180236]: File file.c, Line 418 (ast_openstream): File demo-abouttotry does not exist in any format > WARNING[180236]: File file.c, Line 561 (ast_streamfile): Unable to open demo-abouttotry (format 64): Success > WARNING[180236]: File app_playback.c, Line 83 (playback_exec): ast_streamfile failed on OSS/dsp for demo-abouttotry > WARNING[180236]: File pbx.c, L...

...ction not seem to like the IAX side of things, When I know that the IAX side is functioning great when used from a SIP Phone? I don't know what details would be pertinent to this, but here is what the Asterisk Console Displays: -- Executing Playback("SIP/200-ad26", "demo-abouttotry") in new stack -- Playing 'demo-abouttotry' (language 'en') -- Executing Dial("SIP/200-ad26", "IAX2/guest@misery.digium.com/s@default") in new stack -- Called guest@misery.digium.com/s@default -- Call accepted by 216.207.245.8 (format gsm...

...on frdora core 4. i have two sip UA. if i put canreinvite=yes voice Ok on both sides. and if i change canreinvite=no there is no voice (media through asterisk) one thing more if i try to use playback application for playing some sound file it is also working (like exten => 500,1,Playback(demo-abouttotry) this is working). here is sip.conf //////sip.conf////////////// //////////////// [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes allow=all nat=no [6000] type=peer host=dynamic context=default canreinvite=yes allow=all [1000] type=peer host=dynamic secre...

Greetings again all, With the help of another list member I was able to get the my TDM400P card working properly (and the PRI card loaded too for channels 1-24).

...ew H.323 Connection created. -- Received SETUP message... == Setting up Call -- Calling party name: [5001,] -- Calling party number: [5001] -- Called party name: [500] -- Called party number: [500] -- Executing Playback("H323/ip$10.0.1.219:2303/8", "demo-abouttotry") in new stack -- Playing 'demo-abouttotry' (language 'en') =*= In CreateRealTimeLogicalChannel for call 8 -- externalIpAddress: 10.0.1.207 -- externalPort: 15210 -- SessionID: 1 -- Direction: IsTransmitter -- Started logical channel: sending G.711-uLaw-64k{sw}...

...owing messages. I am behind a NAT server and did NOT change anything in any of the sample config files from CVS. Could this be the problem? BTW - I'm using the Xlite soft phone running on the same box as the asterisk server. -- Executing Playback("SIP/xlite1-625c", "demo-abouttotry") in new stack -- Playing 'demo-abouttotry' (language 'en') -- Executing Dial("SIP/xlite1-625c", "IAX2/guest@misery.digium.com/s@default") in new stack -- Called guest@misery.digium.com/s@default -- IAX2/216.207.245.8:4569-1 is circuit-bu...

...stream BT100 connected to asterisk. Working fine, for calling to each other, and to call via a IAX-Link to the outside. If I try to call the initial demo from the samples.extensions.conf I have nothing to hear. The CLI fine reports: -- Executing Playback("SIP/2209-0260", "demo-abouttotry") in new stack -- Playing 'demo-abouttotry' (language 'en') after a few seconds, when I give it up.... == Spawn extension (demo, 500, 1) exited non-zero on 'SIP/2209-0260' When I call to the voicemail-system with extension 8500, I got also only silence on the ph...

...e SIP proxy in XLite to be the external interface on the firewall, and am able to log into the proxy without difficulty. And while I can begin conversations, I can't keep them going for long. For instance, when trying to call 500@10.0.0.253 (or 500@FWpublicIP), I get most of the "demo-abouttotry" message - "I am about to attempt an IAX connection to a demonstration server located at Di" - at which point it gets cut off. The console spits out the following error: File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call FB9CEC48-7CE1-4171-895B-2DF048ED5D1...

...UTHENTICATED call from 10.0.1.5: > requested format = gsm, > requested prefs = (), > actual format = ulaw, > host prefs = (g729|ulaw|gsm|alaw|ilbc), > priority = mine -- Executing [500@autocontext:1] Playback("IAX2/2002-3", "demo-abouttotry") in new stack -- Playing 'demo-abouttotry' (language 'en') -- Executing [500@autocontext:2] Dial("IAX2/2002-3", "IAX2/guest@misery.digium.com/s@default") in new stack -- Called guest@misery.digium.com/s@default -- Call accepted by 216.207.2...

Hi, i've just got 2 grandstream phones and when I try to connect them with * I get the following: -- Playing 'demo-abouttotry' (language 'en') WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 02da8d3d-a8de-c7e1-5f07-c032ab23d909@10.1.1.179 for seqno 59134 (Response) I've seen there was some discussion on this already but i couldn't find any resolution. Can an...

...'t hear anything. So for example the "500", at the console it shows that is running the playback function with the correct .gsm file, but it never goes on to the next step. I'm running asterisk with "-dvvvvvvgc" params and the console just shows: -- Playing 'demo-abouttotry' (language 'en') So I ran /etc/init.d/zaptal stop and restarted Asterisk, then I tried accessing the "500" extension again. Presto, it works fine. ??? Note, with the Zaptal driver installed I can still connect between the two X-lite phones, no problem. So I'm rather c...

...ost when that happens. ; ; XXX Not yet implemented XXX ; static=yes ; ; if static=yes and writeprotect=no, you can save dialplan by ; CLI command 'save dialplan' too ; writeprotect=no [default] exten => 100,1,Dial,SIP/100 exten => 200,1,Dial,SIP/200 exten => 500,1,Playback(demo-abouttotry); Let them know what's going on exten => 500,2,Dial(IAX2/guest@misery.digium.com/s@default) ; Call the Asterisk demo exten => 500,3,Playback(demo-nogo) ; Couldn't connect to the demo site exten => 500,4,Goto(s,6) ; Return to the start over message. ;exten => 2382031,1,Playback...

...lows [2203] type=friend username=2203 secret=2203 host=dynamic defaultip=192.168.0.2 dtmfmode=inband canreinvite=yes the console on * when running with -vvvvc says :- (whenb trying to dial extension 500 (the demo) from my xten sipphone) -- Executing Playback("SIP/2203-2e99", "demo-abouttotry") in new stack -- Playing 'demo-abouttotry' (language 'en') WARNING[81926]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceede d on call A782222F-BD89-41F0-96CF-4CEA43C20C30@192.168.0.2 for seqno 50218 (Resp onse) WARNING[262160]: File file.c, Line 512 (ast_re...

...AILBOX}) exten => ${MAILBOX},1,VoiceMailMain(); exten => time,1,Answer exten => time,2,Playback,current-time [voice] ;Voicemail System exen => 999,1,Voicemail2 [meeting] exten => 8600,1,Meetme [hold] exten => 6600,1,WaitMusicOnHold() [demo] exten => 500,1,Playback(demo-abouttotry); Let them know what's going on exten => 500,2,Dial(IAX2/guest@misery.digium.com/s@default) ; Call the Asterisk demo exten => 500,3,Playback(demo-nogo) ; Couldn't connect to the demo site exten => 500,4,Goto(s,6) ; Return to the start over message. [defaul...

Hello configuration as follows, dial-out works: capi.conf: [hfcpci] ;;PointToPoint (55512-0) isdnmode=MSN incomingmsn=* ;msn=61 controller=1 devices=2 context=incoming extensions.conf: [incoming] exten => _XX,1,Playback(demo-abouttotry) exten => _XX,n,Dial,SIP/xlite1 exten => _XX,n,HangUp When call is placed, the following debug info is shown, after the last line, it stalls until caller gives up: INFO_IND ID=001 #0x040a LEN=0016 Controller/PLCI/NCCI = 0x101 InfoNumber = 0x7e InfoEle...

...ll-ID: 2143389df4877360@64.7.189.14 CSeq: 19607 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: <sip:500@64.7.161.26> Content-Length: 0 --- -- Executing Playback("SIP/budgeTone-PubIP-7e44", "demo-abouttotry") in new stack We're at 64.7.161.26 port 13648 Adding codec 0x4 (ulaw) to SDP Adding codec 0x400 (ilbc) to SDP Reliably Transmitting (no NAT) to 64.7.189.14:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bK1b2220ace977c3a7;received=64.7.189.14 From: "Budge Tone"...

I've built the latest CVS of asterisk -- not the zaptel or libpri directories, just the asterisk directory. asterisk installs successfully, but there are severe problems. I built this system in the past and ran it, but now building it again fails. This is the CVS as of this morning, 2003-06-13, but I had problems on 06-11/12 as well. After make; make install; make samples; make config, I

Is it possible to execute a script upon successful registration and authentication of a SIP device in Asterisk? For instance, have a script log all successful registrations in a database or authenticate users instead of using the secret=password in the sip.conf file? Thanks - -- Brian Wilkins Software Engineer brian@hcc.net Heritage Communications Corporation Melbourne, FL USA

...102,Voicemail(b4002) > exten => 4002,103,Hangup > > exten => 4003,1,Dial(ZAP/3,30) > ;exten => 4003,2,Voicemail(u4003) > ;exten => 4003,102,Voicemail(b4003) > exten => 4003,103,Hangup > > exten => 2050,1,SetLanguage(en) > exten => 2050,2,Playback(demo-abouttotry) > exten => 2050,3,Dial(IAX2/guest@misery.digium.com/s@default) > exten => 2050,4,Playback(demo-nogo) > exten => 2050,5,Hangup > > > PS- I have the voicemail lines commented out for extensions 4001-4003 > because I haven't set them up yet. >

...; If they press #, return to start exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain [default] exten => 500,1,Playback(demo-abouttotry); Let them know what's going on exten => 500,2,Dial(IAX2/guest@misery.digium.com/s@default) ; Call the Asterisk demo exten => 500,3,Playback(demo-nogo) ; Couldn't connect to the demo site exten => 500,4,Goto(s,6) ; Return to the start over message. [sip-unauthoriz...