Sorry for all the newbie questions. I really appreciate everyone's help today. Okay I've got outgoing and incoming calls working with no echo. yay! Now I'm having an issue with SIP extension to extension calling. Any time I dial another extension it goes right into voice mail. My extensions.conf is pretty small and rough but, here's what I have right now. Most of it was taken from the voip-info website. Any help as always VERY appreciated. Thanks again! Nora Lavelle # cat extensions.conf [incoming] exten => s,1,Answer(); exten => s,2,Background(ssn-greeting); exten => *,1,Directory(default) exten => 205,1,Wait(2) exten => 205,2,Record(/tmp/asterisk-recording:gsm) exten => 205,3,Wait(2) exten => 205,4,Playback(/tmp/asterisk-recording) exten => 205,5,Wait(2) exten => 205,6,Hangup [internal] exten => 101,1,Macro(stdexten,SIP/101) exten => 102,1,Macro(stdexten,SIP/102) exten => 103,1,Macro(stdexten,SIP/103) exten => 123,1,Macro(stdexten,SIP/123) exten => 124,1,Macro(stdexten,SIP/124) exten => 125,1,Macro(stdexten,SIP/125) exten => 126,1,Macro(stdexten,SIP/126) exten => 127,1,Macro(stdexten,SIP/127) exten => 128,1,Macro(stdexten,SIP/128) exten => 129,1,Macro(stdexten,SIP/129) exten => 130,1,Macro(stdexten,SIP/130) exten => 135,1,Macro(stdexten,SIP/135) exten => 117,1,Macro(stdexten,SIP/117) exten => 201,1,Macro(stdexten,SIP/201) [voicemail] exten => 300,1,Answer exten => 300,2,VoicemailMain(ssn-voicemail-greeting) exten => 300,3,Hangup [local] exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) exten => _9NXXXXXX,2,Congestion [longdistance] exten => _91NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) exten => _91NXXNXXXXXX,2,Congestion [macro-stdexten] exten => s,1,Dial(${ARG1},20) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}) exten => s-NOANSWER,2,Goto(default,s,1) exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN}) exten => s-BUSY,2,Goto(default,s,1) exten => s-.,1,Goto(s-NOANSWER,1) exten => a,1,VoicemailMain(${MACRO_EXTEN}) [default] include => incoming include => internal include => voicemail include => local include => longdistance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060126/7e527cdf/attachment.htm
Check your error messages in you asterisk console. Perhaps your sip secret or caller id is broken? What type of phones are you using? On 1/26/06, Nora Lavelle <nora@silverspringnet.com> wrote:> > > > Sorry for all the newbie questions. I really appreciate everyone's help > today. > > > > Okay I've got outgoing and incoming calls working with no echo. yay! Now I'm > having an issue with SIP extension to extension calling. Any time I dial > another extension it goes right into voice mail. My extensions.conf is > pretty small and rough but, here's what I have right now. Most of it was > taken from the voip-info website. Any help as always VERY appreciated. > > > > Thanks again! > > Nora Lavelle > > > > # cat extensions.conf > > [incoming] > > exten => s,1,Answer(); > > exten => s,2,Background(ssn-greeting); > > exten => *,1,Directory(default) > > exten => 205,1,Wait(2) > > exten => 205,2,Record(/tmp/asterisk-recording:gsm) > > exten => 205,3,Wait(2) > > exten => 205,4,Playback(/tmp/asterisk-recording) > > exten => 205,5,Wait(2) > > exten => 205,6,Hangup > > > > [internal] > > exten => 101,1,Macro(stdexten,SIP/101) > > exten => 102,1,Macro(stdexten,SIP/102) > > exten => 103,1,Macro(stdexten,SIP/103) > > exten => 123,1,Macro(stdexten,SIP/123) > > exten => 124,1,Macro(stdexten,SIP/124) > > exten => 125,1,Macro(stdexten,SIP/125) > > exten => 126,1,Macro(stdexten,SIP/126) > > exten => 127,1,Macro(stdexten,SIP/127) > > exten => 128,1,Macro(stdexten,SIP/128) > > exten => 129,1,Macro(stdexten,SIP/129) > > exten => 130,1,Macro(stdexten,SIP/130) > > exten => 135,1,Macro(stdexten,SIP/135) > > exten => 117,1,Macro(stdexten,SIP/117) > > exten => 201,1,Macro(stdexten,SIP/201) > > > > [voicemail] > > exten => 300,1,Answer > > exten => 300,2,VoicemailMain(ssn-voicemail-greeting) > > exten => 300,3,Hangup > > > > [local] > > exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) > > exten => _9NXXXXXX,2,Congestion > > > > [longdistance] > > exten => _91NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) > > exten => _91NXXNXXXXXX,2,Congestion > > > > [macro-stdexten] > > exten => s,1,Dial(${ARG1},20) > > exten => s,2,Goto(s-${DIALSTATUS},1) > > exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}) > > exten => s-NOANSWER,2,Goto(default,s,1) > > exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN}) > > exten => s-BUSY,2,Goto(default,s,1) > > exten => s-.,1,Goto(s-NOANSWER,1) > > exten => a,1,VoicemailMain(${MACRO_EXTEN}) > > > > [default] > > include => incoming > > include => internal > > include => voicemail > > include => local > > include => longdistance > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > >
Here's what I get in the the log this is when extension 130 dials extension 129. Thanks again ! nora -- Executing Macro("SIP/130-a644", "stdexten|SIP/129") in new stack -- Executing Dial("SIP/130-a644", "SIP/129|20") in new stack -- Called 129 Jan 26 17:20:48 WARNING[28243]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 4c6f199b2f02a5e05ff21c140eade0a1@10.200.1.234 for seqno 102 (Critical Request) == No one is available to answer at this time -- Executing Goto("SIP/130-a644", "s-NOANSWER|1") in new stack -- Goto (macro-stdexten,s-NOANSWER,1) -- Executing VoiceMail("SIP/130-a644", "u129") in new stack -- Playing 'voicemail/default/129/unavail' (language 'en') == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'SIP/130-a644' in macro 'stdexten' == Spawn extension (default, 129, 1) exited non-zero on 'SIP/130-a644' -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gary Richardson Sent: Thursday, January 26, 2006 5:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] extension to extension dialing Check your error messages in you asterisk console. Perhaps your sip secret or caller id is broken? What type of phones are you using? On 1/26/06, Nora Lavelle <nora@silverspringnet.com> wrote:> > > > Sorry for all the newbie questions. I really appreciate everyone'shelp> today. > > > > Okay I've got outgoing and incoming calls working with no echo. yay!Now I'm> having an issue with SIP extension to extension calling. Any time Idial> another extension it goes right into voice mail. My extensions.confis> pretty small and rough but, here's what I have right now. Most of itwas> taken from the voip-info website. Any help as always VERY appreciated. > > > > Thanks again! > > Nora Lavelle > > > > # cat extensions.conf > > [incoming] > > exten => s,1,Answer(); > > exten => s,2,Background(ssn-greeting); > > exten => *,1,Directory(default) > > exten => 205,1,Wait(2) > > exten => 205,2,Record(/tmp/asterisk-recording:gsm) > > exten => 205,3,Wait(2) > > exten => 205,4,Playback(/tmp/asterisk-recording) > > exten => 205,5,Wait(2) > > exten => 205,6,Hangup > > > > [internal] > > exten => 101,1,Macro(stdexten,SIP/101) > > exten => 102,1,Macro(stdexten,SIP/102) > > exten => 103,1,Macro(stdexten,SIP/103) > > exten => 123,1,Macro(stdexten,SIP/123) > > exten => 124,1,Macro(stdexten,SIP/124) > > exten => 125,1,Macro(stdexten,SIP/125) > > exten => 126,1,Macro(stdexten,SIP/126) > > exten => 127,1,Macro(stdexten,SIP/127) > > exten => 128,1,Macro(stdexten,SIP/128) > > exten => 129,1,Macro(stdexten,SIP/129) > > exten => 130,1,Macro(stdexten,SIP/130) > > exten => 135,1,Macro(stdexten,SIP/135) > > exten => 117,1,Macro(stdexten,SIP/117) > > exten => 201,1,Macro(stdexten,SIP/201) > > > > [voicemail] > > exten => 300,1,Answer > > exten => 300,2,VoicemailMain(ssn-voicemail-greeting) > > exten => 300,3,Hangup > > > > [local] > > exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) > > exten => _9NXXXXXX,2,Congestion > > > > [longdistance] > > exten => _91NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) > > exten => _91NXXNXXXXXX,2,Congestion > > > > [macro-stdexten] > > exten => s,1,Dial(${ARG1},20) > > exten => s,2,Goto(s-${DIALSTATUS},1) > > exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}) > > exten => s-NOANSWER,2,Goto(default,s,1) > > exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN}) > > exten => s-BUSY,2,Goto(default,s,1) > > exten => s-.,1,Goto(s-NOANSWER,1) > > exten => a,1,VoicemailMain(${MACRO_EXTEN}) > > > > [default] > > include => incoming > > include => internal > > include => voicemail > > include => local > > include => longdistance > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > >_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Snom's don't care, port 2057 is fine. Can you ping each phone from the Linux console? -----Original Message----- From: Gary Richardson [mailto:gary.richardson@gmail.com] Sent: Thursday, January 26, 2006 7:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] extension to extension dialing In your sip.conf, make sure these phones have a Type=Friend entry and a qualify=yes. I don't think the qualify=yes is required, but it helps in debuging. About the port, I'm not too sure about sipura and snom phones (I only have Cisco phones :(). That could have something to do with it.. On 1/26/06, Nora Lavelle <nora@silverspringnet.com> wrote:> > Hi there gary. thanks so much for your help. we're using sipura-841 andsnom 320s.> > Here's the sip show peers.. that's weird that extension 130 has port2057.. could that be the problem ?> > -nora > > Name/username Host Dyn Nat ACL Mask PortStatus> > 201/201 10.200.0.56 D 255.255.255.255 5060Unmonitor> ed > 130/130 10.200.0.10 D 255.255.255.255 2057Unmonitor> ed > 129/129 10.200.0.5 D 255.255.255.255 5060Unmonitor> ed > 127/127 10.201.0.30 D 255.255.255.255 5060Unmonitor> ed > 126/126 10.201.0.29 D 255.255.255.255 5060Unmonitor> ed > 125/125 10.201.0.35 D 255.255.255.255 5060Unmonitor> ed > 124/124 10.201.0.31 D 255.255.255.255 5060Unmonitor> ed > 102/102 10.200.0.48 D 255.255.255.255 5060Unmonitor> ed > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com on behalf of Gary Richardson > Sent: Thu 1/26/2006 5:18 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] extension to extension dialing > > Check your error messages in you asterisk console. Perhaps your sip > secret or caller id is broken? > > What type of phones are you using? > > On 1/26/06, Nora Lavelle <nora@silverspringnet.com> wrote: > > > > > > > > Sorry for all the newbie questions. I really appreciate everyone's help > > today. > > > > > > > > Okay I've got outgoing and incoming calls working with no echo. yay! NowI'm> > having an issue with SIP extension to extension calling. Any time I dial > > another extension it goes right into voice mail. My extensions.conf is > > pretty small and rough but, here's what I have right now. Most of it was > > taken from the voip-info website. Any help as always VERY appreciated. > > > > > > > > Thanks again! > > > > Nora Lavelle > > > > > > > > # cat extensions.conf > > > > [incoming] > > > > exten => s,1,Answer(); > > > > exten => s,2,Background(ssn-greeting); > > > > exten => *,1,Directory(default) > > > > exten => 205,1,Wait(2) > > > > exten => 205,2,Record(/tmp/asterisk-recording:gsm) > > > > exten => 205,3,Wait(2) > > > > exten => 205,4,Playback(/tmp/asterisk-recording) > > > > exten => 205,5,Wait(2) > > > > exten => 205,6,Hangup > > > > > > > > [internal] > > > > exten => 101,1,Macro(stdexten,SIP/101) > > > > exten => 102,1,Macro(stdexten,SIP/102) > > > > exten => 103,1,Macro(stdexten,SIP/103) > > > > exten => 123,1,Macro(stdexten,SIP/123) > > > > exten => 124,1,Macro(stdexten,SIP/124) > > > > exten => 125,1,Macro(stdexten,SIP/125) > > > > exten => 126,1,Macro(stdexten,SIP/126) > > > > exten => 127,1,Macro(stdexten,SIP/127) > > > > exten => 128,1,Macro(stdexten,SIP/128) > > > > exten => 129,1,Macro(stdexten,SIP/129) > > > > exten => 130,1,Macro(stdexten,SIP/130) > > > > exten => 135,1,Macro(stdexten,SIP/135) > > > > exten => 117,1,Macro(stdexten,SIP/117) > > > > exten => 201,1,Macro(stdexten,SIP/201) > > > > > > > > [voicemail] > > > > exten => 300,1,Answer > > > > exten => 300,2,VoicemailMain(ssn-voicemail-greeting) > > > > exten => 300,3,Hangup > > > > > > > > [local] > > > > exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) > > > > exten => _9NXXXXXX,2,Congestion > > > > > > > > [longdistance] > > > > exten => _91NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) > > > > exten => _91NXXNXXXXXX,2,Congestion > > > > > > > > [macro-stdexten] > > > > exten => s,1,Dial(${ARG1},20) > > > > exten => s,2,Goto(s-${DIALSTATUS},1) > > > > exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}) > > > > exten => s-NOANSWER,2,Goto(default,s,1) > > > > exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN}) > > > > exten => s-BUSY,2,Goto(default,s,1) > > > > exten => s-.,1,Goto(s-NOANSWER,1) > > > > exten => a,1,VoicemailMain(${MACRO_EXTEN}) > > > > > > > > [default] > > > > include => incoming > > > > include => internal > > > > include => voicemail > > > > include => local > > > > include => longdistance > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Hmm.. I definitely have type=friend in the sip.conf and I added qualify=yes but, I think that's default anyways.. When I call from the outside and enter his extension it goes through to him fine but, when I go extension to extension it automatically goes to voicemail.. Here are the messages from the console: -- Executing Macro("SIP/130-58df", "stdexten|SIP/124") in new stack -- Executing Dial("SIP/130-58df", "SIP/124|20") in new stack -- Called 124 Jan 27 10:27:10 WARNING[28243]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 7998a5e87b708f4374ca0ec212863b6d@10.200.1.234 for seqno 102 (Critical Request) == No one is available to answer at this time -- Executing Goto("SIP/130-58df", "s-NOANSWER|1") in new stack -- Goto (macro-stdexten,s-NOANSWER,1) -- Executing VoiceMail("SIP/130-58df", "u124") in new stack -- Playing 'voicemail/default/124/greet' (language 'en') Jan 27 10:27:10 NOTICE[28243]: sched.c:290 ast_sched_del: Attempted to delete non-existant schedule entry 22838! -- Playing 'vm-isunavail' (language 'en') -- Playing 'vm-intro' (language 'en') -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gary Richardson Sent: Thursday, January 26, 2006 6:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] extension to extension dialing In your sip.conf, make sure these phones have a Type=Friend entry and a qualify=yes. I don't think the qualify=yes is required, but it helps in debuging. About the port, I'm not too sure about sipura and snom phones (I only have Cisco phones :(). That could have something to do with it.. On 1/26/06, Nora Lavelle <nora@silverspringnet.com> wrote:> > Hi there gary. thanks so much for your help. we're using sipura-841and snom 320s.> > Here's the sip show peers.. that's weird that extension 130 has port2057.. could that be the problem ?> > -nora > > Name/username Host Dyn Nat ACL Mask PortStatus> > 201/201 10.200.0.56 D 255.255.255.255 5060Unmonitor> ed > 130/130 10.200.0.10 D 255.255.255.255 2057Unmonitor> ed > 129/129 10.200.0.5 D 255.255.255.255 5060Unmonitor> ed > 127/127 10.201.0.30 D 255.255.255.255 5060Unmonitor> ed > 126/126 10.201.0.29 D 255.255.255.255 5060Unmonitor> ed > 125/125 10.201.0.35 D 255.255.255.255 5060Unmonitor> ed > 124/124 10.201.0.31 D 255.255.255.255 5060Unmonitor> ed > 102/102 10.200.0.48 D 255.255.255.255 5060Unmonitor> ed > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com on behalf of GaryRichardson> Sent: Thu 1/26/2006 5:18 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] extension to extension dialing > > Check your error messages in you asterisk console. Perhaps your sip > secret or caller id is broken? > > What type of phones are you using? > > On 1/26/06, Nora Lavelle <nora@silverspringnet.com> wrote: > > > > > > > > Sorry for all the newbie questions. I really appreciate everyone'shelp> > today. > > > > > > > > Okay I've got outgoing and incoming calls working with no echo. yay!Now I'm> > having an issue with SIP extension to extension calling. Any time Idial> > another extension it goes right into voice mail. My extensions.confis> > pretty small and rough but, here's what I have right now. Most of itwas> > taken from the voip-info website. Any help as always VERYappreciated.> > > > > > > > Thanks again! > > > > Nora Lavelle > > > > > > > > # cat extensions.conf > > > > [incoming] > > > > exten => s,1,Answer(); > > > > exten => s,2,Background(ssn-greeting); > > > > exten => *,1,Directory(default) > > > > exten => 205,1,Wait(2) > > > > exten => 205,2,Record(/tmp/asterisk-recording:gsm) > > > > exten => 205,3,Wait(2) > > > > exten => 205,4,Playback(/tmp/asterisk-recording) > > > > exten => 205,5,Wait(2) > > > > exten => 205,6,Hangup > > > > > > > > [internal] > > > > exten => 101,1,Macro(stdexten,SIP/101) > > > > exten => 102,1,Macro(stdexten,SIP/102) > > > > exten => 103,1,Macro(stdexten,SIP/103) > > > > exten => 123,1,Macro(stdexten,SIP/123) > > > > exten => 124,1,Macro(stdexten,SIP/124) > > > > exten => 125,1,Macro(stdexten,SIP/125) > > > > exten => 126,1,Macro(stdexten,SIP/126) > > > > exten => 127,1,Macro(stdexten,SIP/127) > > > > exten => 128,1,Macro(stdexten,SIP/128) > > > > exten => 129,1,Macro(stdexten,SIP/129) > > > > exten => 130,1,Macro(stdexten,SIP/130) > > > > exten => 135,1,Macro(stdexten,SIP/135) > > > > exten => 117,1,Macro(stdexten,SIP/117) > > > > exten => 201,1,Macro(stdexten,SIP/201) > > > > > > > > [voicemail] > > > > exten => 300,1,Answer > > > > exten => 300,2,VoicemailMain(ssn-voicemail-greeting) > > > > exten => 300,3,Hangup > > > > > > > > [local] > > > > exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) > > > > exten => _9NXXXXXX,2,Congestion > > > > > > > > [longdistance] > > > > exten => _91NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) > > > > exten => _91NXXNXXXXXX,2,Congestion > > > > > > > > [macro-stdexten] > > > > exten => s,1,Dial(${ARG1},20) > > > > exten => s,2,Goto(s-${DIALSTATUS},1) > > > > exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}) > > > > exten => s-NOANSWER,2,Goto(default,s,1) > > > > exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN}) > > > > exten => s-BUSY,2,Goto(default,s,1) > > > > exten => s-.,1,Goto(s-NOANSWER,1) > > > > exten => a,1,VoicemailMain(${MACRO_EXTEN}) > > > > > > > > [default] > > > > include => incoming > > > > include => internal > > > > include => voicemail > > > > include => local > > > > include => longdistance > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users