On 9 Dec 2010 at 13:31, Gary (Gary Kuznitz <[email protected]>) commented about [asterisk-users] (Fwd) Re: Configuring Softphone:
> Thank you for the reply. > > On 8 Dec 2010 at 13:38, Danny (Danny Nicholas <[email protected]>) commented > about RE: [asterisk-users] Configuring Softphone: > > > -----Original Message----- > > From: [email protected] > > [mailto:[email protected]] On Behalf Of Gary Kuznitz > > Sent: Wednesday, December 08, 2010 1:27 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [asterisk-users] Configuring Softphone > > > > The phone is finally registering. That's great. > > > > I'm trying to understand what all these lines in Extensions.conf are > > defining. > > I can't make heads or tails them. I have been reading the manual > > AsteriskManualTheFutureOfTelephony2ndEdition. > > > > I'm currently getting an error when placing a call on the cmd line saying: > > NOTICE[5630]: chan_sip.c:14383 handle_request_invite: Call from 'Gary' to > > extension '91AreaCodePhone#' rejected because extension not found. > > > > > > What I have in Extensions.conf is: > > [gary-incomming] > > exten => 1001,1,Dial(IAX2/gogh) > > exten => 1001,2,HangUp() > > exten => 120,1,Dial(SIP/Gary) > > exten => Gary,1,Goto(120,1) > > exten => i,1,Playback(invalid) > > exten => i,2,Goto(s,1) > > exten => s,1,Wait(1) > > exten => s,2,Answer() > > exten => s,3,NoOp(${CALLERID}) > > exten => s,4,NoOp(${CALLERIDNUM}) > > exten => s,5,NoOp(${CALLERIDNAME}) > > exten => s,6,Wait(4) > > exten => s,7,Playback(vm-goodbye) > > exten => s,8,Wait(2) > > exten => s,9,HangUp() > > > > What I have in Sip.conf is: > > [authentication] > > > > [general] > > context = default > > allowoverlap = no > > bindport = 5060 > > bindaddr = 0.0.0.0 > > srvlookup = yes > > limitonpeers = yes > > allowguest=no > > nat=yes > > > > [Gary] > > type = friend > > username = Gary > > callerid = 120 > > secret = password > > host = dynamic > > defaultip = dynamic > > context = gary-incomming > > dtmfmode = rfc2833 > > allow=all > > > > Frustrated, > > > > Gary > > > > Without any other comment, you need > > exten => _91.,1,Dial(DAHDI/g1/${EXTEN}) > > in the gary-incomming context. > > > > As defined now, Gary can > > #1 answer a call > > #2 call IAX/gogh using 1001 > > > > I entered the exten line you suggested: > [gary-incomming] > exten => _91.,1,Dial(DAHDI/g1/${EXTEN}) > > I removed all other lines in [gary-incomming] > > When I place a call I get on the cmd line: > -- Executing [916618579...@gary-incomming:1] Dial("SIP/Gary-08941b20", > "DAHDI/g1/916618579191") in new stack > -- Called g1/916618579191 > -- DAHDI/1-1 answered SIP/Gary-08941b20 > [Dec 9 14:00:37] WARNING[5630]: chan_sip.c:1958 retrans_pkt: Maximum retries > exceeded on transmission [email protected] for seqno 669 > (Critical Response) -- See doc/sip-retransmit.txt. > [Dec 9 14:00:37] WARNING[5630]: chan_sip.c:1980 retrans_pkt: Hanging up call > [email protected] - no reply to our critical packet (see > doc/sip-retransmit.txt). > -- Hungup 'DAHDI/1-1' > == Spawn extension (gary-incomming, 916618579191, 1) exited non-zero on > 'SIP/Gary-08941b20' > > Do you have any ideas? Would you like to see what is in extensions.conf for > a local > extension? > > Thank you, > > Gary I'm getting closer. Express Talk is now making the call. I'm getting an error on the cmd line: -- Executing [91myareacodepho...@dlpn_dialplan1:1] Macro("SIP/120- b6003810", "trunkdial-failover-0.3|Dahdi/g1/1MyAreaCodePhone#||trunk_1|") in new stack -- Executing [[email protected]:1] GotoIf("SIP/120-b6003810", "0?1- fmsetcid|1") in new stack -- Executing [[email protected]:2] GotoIf("SIP/120-b6003810", "0?1- setgbobname|1") in new stack -- Executing [[email protected]:3] Set("SIP/120-b6003810", "CALLERID(num)=") in new stack -- Executing [[email protected]:4] GotoIf("SIP/120-b6003810", "0?1- dial|1") in new stack -- Executing [[email protected]:5] Set("SIP/120-b6003810", "CALLERID(all)=") in new stack -- Executing [[email protected]:6] Goto("SIP/120-b6003810", "1- dial|1") in new stack -- Goto (macro-trunkdial-failover-0.3,1-dial,1) -- Executing [[email protected]:1] Dial("SIP/120-b6003810", "Dahdi/g1/1MyAreaCodePhone#") in new stack -- Called g1/1MyAreaCodePhone# -- DAHDI/1-1 answered SIP/120-b6003810 -- Hungup 'DAHDI/1-1' == Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero on 'SIP/120-b6003810' in macro 'trunkdial-failover-0.3' == Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero on 'SIP/120-b6003810' [Dec 9 18:40:39] WARNING[5806]: chan_sip.c:1958 retrans_pkt: Maximum retries exceeded on transmission [email protected] for seqno 287 (Critical Response) -- See doc/sip-retransmit.txt. I don't know if this has anything to do with Express Talk using Local RTP ports to listen 8000-8020 and Asterisk using 10000 and up. I tried changing Express Talk to 10000-10020 and forwarding those to Asterisk. It didn't seam to help. I currently have in extensions.conf: [gary-incomming] exten => s,1,Wait(1) exten => s,2,Answer() exten => s,3,NoOp(${CALLERID}) exten => s,n,NoOp(${CALLERIDNUM}) exten => s,n,NoOp(${CALLERIDNAME}) exten => s,n,Wait(4) exten => s,n,Playback(tt-weasels) exten => s,n,Voicemail(11...@vm-test) exten => s,n,Wait(2) exten => s,n,Playback(vm-goodbye) exten => s,n,Wait(2) exten => s,n,HandUp() exten => 120,1,Dial(SIP/gary) exten => gary,1,Goto(120,1) exten => i,1,Playback(invalid) exten => i,2,Goto(s,1) There are some other issues but I thought I should pose one question at a time. Could someone please give me an idea as to why I'm getting the warning? Thank you, Gary -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
