Re: [asterisk-users] (Fwd) Re: Configuring Softphone
On Thu, 9 Dec 2010, Gary Kuznitz wrote: > 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 [...@macro-trunkdial-failover-0.3:1] > GotoIf("SIP/120-b6003810", "0?1- > fmsetcid|1") in new stack >-- Executing [...@macro-trunkdial-failover-0.3:2] > GotoIf("SIP/120-b6003810", "0?1- > setgbobname|1") in new stack >-- Executing [...@macro-trunkdial-failover-0.3:3] Set("SIP/120-b6003810", > "CALLERID(num)=") in new stack >-- Executing [...@macro-trunkdial-failover-0.3:4] > GotoIf("SIP/120-b6003810", "0?1- > dial|1") in new stack >-- Executing [...@macro-trunkdial-failover-0.3:5] Set("SIP/120-b6003810", > "CALLERID(all)=") in new stack >-- Executing [...@macro-trunkdial-failover-0.3:6] Goto("SIP/120-b6003810", > "1- > dial|1") in new stack >-- Goto (macro-trunkdial-failover-0.3,1-dial,1) >-- Executing [1-d...@macro-trunkdial-failover-0.3:1] > 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 1291829922-5076-gar...@192.168.168.7 for seqno 287 > (Critical Response) -- See doc/sip-retransmit.txt. > 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) Does it seem odd that your console output does not match your dialplan? I would suggest discarding PIAF or Elastix or whatever created your dialplan and start from scratch. Once you master the concepts and interaction between sip.conf and extensions.conf you will be in a better place to evaluate the merits of using a GUI to create your dialplan or continue growing your own. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
Re: [asterisk-users] (Fwd) Re: Configuring Softphone
On 9 Dec 2010 at 13:31, Gary (Gary Kuznitz ) commented about [asterisk-users] (Fwd) Re: Configuring Softphone: > Thank you for the reply. > > On 8 Dec 2010 at 13:38, Danny (Danny Nicholas ) commented > about RE: [asterisk-users] Configuring Softphone: > > > -Original Message- > > From: asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] 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 1291829914-5076-gar...@192.168.168.7 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 > 1291829914-5076-gar...@192.168.168.7 - 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 [...@macro-trunkdial-failover-0.3:1] GotoIf("SIP/120-b6003810", "0?1- fmsetcid|1") in new stack -- Executing [...@macro-trunkdial-failover-0.3:2] GotoIf("SIP/120-b6003810", "0?1- setgbobname|1") in new stack -- Executing [...@macro-trunkdial-failover-0.3:3] Set("SIP/120-b6003810", "CALLERID(num)=") in new stack -- Executing [...@macro-trunkdial-failover-0.3:4] GotoIf("SIP/120-b6003810", "0?1- dial|1") in new stack -- Executing [...@macro-trunkdial-failover-0.3:5] Set("SIP/120-b6003810", "CALLERID(all)
[asterisk-users] (Fwd) Re: Configuring Softphone
Thank you for the reply. On 8 Dec 2010 at 13:38, Danny (Danny Nicholas ) commented about RE: [asterisk-users] Configuring Softphone: > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] 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 1291829914-5076-gar...@192.168.168.7 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 1291829914-5076-gar...@192.168.168.7 - 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 --- End of forwarded message --- WPM$44FF.PM$ Description: Mail message body -- _ -- 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