Hi I looked at few emails related to this subject. And still not sure
how to solve the loop detect problem for my case
iqbala at improvise:/etc/asterisk$ cat sip.conf
[general]
context=line1
[phone]
type=friend
context=phone1
secret=g00dpazzwerd
bindport=5060
host=192.168.1.106
dtmfmode=rfc2833
[line]
type=friend
context=line1
secret=anothers33cret
bindport=5061
host=192.168.1.106
dtmfmode=rfc2833
iqbala at improvise:/etc/asterisk$ cat extensions.conf
[default]
exten => s,1,Answer
exten => s,2,Wait(2)
exten => s,3,Playback(tt-monkeys)
exten => s,4,Hangup
[from-internal]
include => default
[phone1]
[from-pstn]
;include => default
exten => s,1,Dial(SIP/phone at phone,10)
exten => s,2,Voicemail(line)
exten => s,3,Hangup
[line1]
So my home land line is going to the FXO port and my home phone is
hanging off of FXS port.
Here are the contexts for my fxo/fxs card
improvise*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret
pseudo default default
1 from-internal default
2 from-internal default
3 from-pstn default
4 from-pstn default
I want to call from my cell and make my home phone ring and if I dont
pickup in 10 secs I want the call
go to my voicemail. But I am getting a loop detect. The debug output
is attached.
What am I doing wrong?
--
Asif Iqbal
PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu
A: Because it messes up the order in which people normally read text.
Q: Why is top-posting such a bad thing?
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On Sun, Mar 15, 2009 at 6:28 PM, Asif Iqbal <vadud3 at gmail.com> wrote:> Hi I looked at few emails related to this subject. And still not sure > how to solve the loop detect problem for my case > > iqbala at improvise:/etc/asterisk$ cat sip.conf > > [general] > context=line1 > > [phone] > type=friend > context=phone1 > secret=g00dpazzwerd > bindport=5060 > host=192.168.1.106 > dtmfmode=rfc2833 > > [line] > type=friend > context=line1 > secret=anothers33cret > bindport=5061 > host=192.168.1.106 > dtmfmode=rfc2833 > > iqbala at improvise:/etc/asterisk$ cat extensions.conf > [default] > exten => s,1,Answer > exten => s,2,Wait(2) > exten => s,3,Playback(tt-monkeys) > exten => s,4,Hangup > > [from-internal] > include => default > > [phone1] > > [from-pstn] > ;include => default > exten => s,1,Dial(SIP/phone at phone,10) > exten => s,2,Voicemail(line) > exten => s,3,Hangup > > [line1] > > > So my home land line is going to the FXO port and my home phone is > hanging off of FXS port. > > Here are the contexts for my fxo/fxs card > > > improvise*CLI> dahdi show channels > ? Chan Extension ?Context ? ? ? ? Language ? MOH Interpret > ?pseudo ? ? ? ? ? ?default ? ? ? ? ? ? ? ? ? ?default > ? ? ?1 ? ? ? ? ? ?from-internal ? ? ? ? ? ? ?default > ? ? ?2 ? ? ? ? ? ?from-internal ? ? ? ? ? ? ?default > ? ? ?3 ? ? ? ? ? ?from-pstn ? ? ? ? ? ? ? ? ?default > ? ? ?4 ? ? ? ? ? ?from-pstn ? ? ? ? ? ? ? ? ?default > > > I want to call from my cell and make my home phone ring and if I dont > pickup in 10 secs I want the call > go to my voicemail. But I am getting a loop detect. The debug output > is attached. > > What am I doing wrong? > > -- > Asif Iqbal > PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu > A: Because it messes up the order in which people normally read text. > Q: Why is top-posting such a bad thing? > > _______________________________________________Am I missing something or is your setup dahdi/zaptel only? What is the SIP stuff for? You are doing all TDM, no VoIP from what I gather. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype)
On Sun, Mar 15, 2009 at 8:28 PM, Steve Totaro <stotaro at asteriskhelpdesk.com> wrote:> On Sun, Mar 15, 2009 at 6:28 PM, Asif Iqbal <vadud3 at gmail.com> wrote: >> Hi I looked at few emails related to this subject. And still not sure >> how to solve the loop detect problem for my case >> >> iqbala at improvise:/etc/asterisk$ cat sip.conf >> >> [general] >> context=line1 >> >> [phone] >> type=friend >> context=phone1 >> secret=g00dpazzwerd >> bindport=5060 >> host=192.168.1.106 >> dtmfmode=rfc2833 >> >> [line] >> type=friend >> context=line1 >> secret=anothers33cret >> bindport=5061 >> host=192.168.1.106 >> dtmfmode=rfc2833 >> >> iqbala at improvise:/etc/asterisk$ cat extensions.conf >> [default] >> exten => s,1,Answer >> exten => s,2,Wait(2) >> exten => s,3,Playback(tt-monkeys) >> exten => s,4,Hangup >> >> [from-internal] >> include => default >> >> [phone1] >> >> [from-pstn] >> ;include => default >> exten => s,1,Dial(SIP/phone at phone,10) >> exten => s,2,Voicemail(line) >> exten => s,3,Hangup >> >> [line1] >> >> >> So my home land line is going to the FXO port and my home phone is >> hanging off of FXS port. >> >> Here are the contexts for my fxo/fxs card >> >> >> improvise*CLI> dahdi show channels >> ? Chan Extension ?Context ? ? ? ? Language ? MOH Interpret >> ?pseudo ? ? ? ? ? ?default ? ? ? ? ? ? ? ? ? ?default >> ? ? ?1 ? ? ? ? ? ?from-internal ? ? ? ? ? ? ?default >> ? ? ?2 ? ? ? ? ? ?from-internal ? ? ? ? ? ? ?default >> ? ? ?3 ? ? ? ? ? ?from-pstn ? ? ? ? ? ? ? ? ?default >> ? ? ?4 ? ? ? ? ? ?from-pstn ? ? ? ? ? ? ? ? ?default >> >> >> I want to call from my cell and make my home phone ring and if I dont >> pickup in 10 secs I want the call >> go to my voicemail. But I am getting a loop detect. The debug output >> is attached. >> >> What am I doing wrong? >> >> -- >> Asif Iqbal >> PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu >> A: Because it messes up the order in which people normally read text. >> Q: Why is top-posting such a bad thing? >> >> _______________________________________________ > > Am I missing something or is your setup dahdi/zaptel only? ?What is > the SIP stuff for? ?You are doing all TDM, no VoIP from what I gather.I am probably missing something, being a newbie. I have a 4 port fxs/fxo (2/2) card. My land line is going to one of the FXO port and my home phone is connected to one of the FXS port. I want to be able to call my phone number from external phone (cell phone) and have my home phone ring. And if I do not pick up the phone in 10 secs I want the voicemail to pickup the call. I do have a dialtone when pick up my phone that is attached to the FXS port of my asterisk server> > -- > Thanks, > Steve Totaro > +18887771888 (Toll Free) > +12409381212 (Cell) > +12024369784 (Skype) > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > ? http://lists.digium.com/mailman/listinfo/asterisk-users >-- Asif Iqbal PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing?
> > I am probably missing something, being a newbie. I have a 4 port > fxs/fxo (2/2) card. > > My land line is going to one of the FXO port and my home phone is connected > to one of the FXS port. > > I want to be able to call my phone number from external phone (cell phone) > and have my home phone ring. And if I do not pick up the phone in 10 secs I want > the voicemail to pickup the call. > > I do have a dialtone when pick up my phone that is attached to the FXS port > of my asterisk server > >A printout from the CLI would be helpful - but I think you have your contexts crossed over. (call from outside hitting internal, instead of from-pstn) PaulH
Hi, problem is that you are saying that phone in sip.conf is at the same ip address of your asterisk box so you are dialing into a loop to your self asterisk box [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 what you need is: [phone] type=friend context=phone1 secret=g00dpazzwerd dtmfmode=rfc2833 host=dynamic ;configuring your codecs (i don't know what else you have configured, just preventing audio for you) disallow=all allow=ulaw allow=alaw allow=gsm Dial sip/phone is enough too.. [from-pstn] ;include => default exten => s,1,Dial(SIP/phone,10) exten => s,2,Voicemail(line) exten => s,3,Hangup hope it helps. don't forget to asterisk reload on cli. Looking forward to hearing from you. cheers -- Marco Mouta On Sun, Mar 15, 2009 at 10:28 PM, Asif Iqbal <vadud3 at gmail.com> wrote:> Hi I looked at few emails related to this subject. And still not sure > how to solve the loop detect problem for my case > > iqbala at improvise:/etc/asterisk$ cat sip.conf > > [general] > context=line1 > > [phone] > type=friend > context=phone1 > secret=g00dpazzwerd > bindport=5060 > host=192.168.1.106 > dtmfmode=rfc2833 > > [line] > type=friend > context=line1 > secret=anothers33cret > bindport=5061 > host=192.168.1.106 > dtmfmode=rfc2833 > > iqbala at improvise:/etc/asterisk$ cat extensions.conf > [default] > exten => s,1,Answer > exten => s,2,Wait(2) > exten => s,3,Playback(tt-monkeys) > exten => s,4,Hangup > > [from-internal] > include => default > > [phone1] > > [from-pstn] > ;include => default > exten => s,1,Dial(SIP/phone at phone,10) > exten => s,2,Voicemail(line) > exten => s,3,Hangup > > [line1] > > > So my home land line is going to the FXO port and my home phone is > hanging off of FXS port. > > Here are the contexts for my fxo/fxs card > > > improvise*CLI> dahdi show channels > ? Chan Extension ?Context ? ? ? ? Language ? MOH Interpret > ?pseudo ? ? ? ? ? ?default ? ? ? ? ? ? ? ? ? ?default > ? ? ?1 ? ? ? ? ? ?from-internal ? ? ? ? ? ? ?default > ? ? ?2 ? ? ? ? ? ?from-internal ? ? ? ? ? ? ?default > ? ? ?3 ? ? ? ? ? ?from-pstn ? ? ? ? ? ? ? ? ?default > ? ? ?4 ? ? ? ? ? ?from-pstn ? ? ? ? ? ? ? ? ?default > > > I want to call from my cell and make my home phone ring and if I dont > pickup in 10 secs I want the call > go to my voicemail. But I am getting a loop detect. The debug output > is attached. > > What am I doing wrong? > > -- > Asif Iqbal > PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu > A: Because it messes up the order in which people normally read text. > Q: Why is top-posting such a bad thing? > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > ? http://lists.digium.com/mailman/listinfo/asterisk-users >
Hello Asif, I have experienced 'loop detected' when the peer where I want to send the calls to, and the asterisk Box have both the same IP address (That would make a loop). Could you please verify? Regards, Asif Iqbal wrote:> Hi I looked at few emails related to this subject. And still not sure > how to solve the loop detect problem for my case > > iqbala at improvise:/etc/asterisk$ cat sip.conf > > [general] > context=line1 > > [phone] > type=friend > context=phone1 > secret=g00dpazzwerd > bindport=5060 > host=192.168.1.106 > dtmfmode=rfc2833 > > [line] > type=friend > context=line1 > secret=anothers33cret > bindport=5061 > host=192.168.1.106 > dtmfmode=rfc2833 > > iqbala at improvise:/etc/asterisk$ cat extensions.conf > [default] > exten => s,1,Answer > exten => s,2,Wait(2) > exten => s,3,Playback(tt-monkeys) > exten => s,4,Hangup > > [from-internal] > include => default > > [phone1] > > [from-pstn] > ;include => default > exten => s,1,Dial(SIP/phone at phone,10) > exten => s,2,Voicemail(line) > exten => s,3,Hangup > > [line1] > > > So my home land line is going to the FXO port and my home phone is > hanging off of FXS port. > > Here are the contexts for my fxo/fxs card > > > improvise*CLI> dahdi show channels > Chan Extension Context Language MOH Interpret > pseudo default default > 1 from-internal default > 2 from-internal default > 3 from-pstn default > 4 from-pstn default > > > I want to call from my cell and make my home phone ring and if I dont > pickup in 10 secs I want the call > go to my voicemail. But I am getting a loop detect. The debug output > is attached. > > What am I doing wrong? > > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Jose P. Espinal http://www.eSlackware.com IRC: [OFTC|FreeNode] Khratos @ #slackware | #asterisk/-doc/-bugs