[asterisk-users] Problem: no such extension 'xx' in context 'default'
Hi Guys, I am not so familiar with asterisk and hope to get help here. I am having now some stupid errors. My goal for the first, is to create a simple pbx with different context. As long as I use only the contex 'default' everything seems to work perfect. Now I tried to add another context i.e 'internal' and the asterisk is complaining for not finding the required extension in the 'default' context. Asterisk schould point this to the internal contex and not default. here my simply config data : sip.conf : [general] port=5060 bindaddr=0.0.0.0 [10] type=friend secret=1234 host=dynamic context=internal [11] type=friend secret=1234 host=dynamic context=internal extensions.conf [default] exten =2,1,Playback(digits/2) ; exten =2,2,Goto(default,10,1) exten=3,1,Playback(pbx-invalid) exten=3,2,Goto(default,4,1) exten=4,1,Playback(vm-goodbye) exten=4,2,Hangup() [internal] exten = 10,1,Dial(SIP/10,10) exten =10,2,Background(vm-nobodyavail) exten = 11,1,Dial(SIP/11,5) exten =11,2,Background(vm-nobodyavail) now when I dial 10, I got the following error : no such extension '10' in context 'default' thanks in advance manfred _ Hol dir 30 kostenlose Emoticons für deinen Windows Live Messenger http://www.livemessenger-emoticons.com/funfamily/de-at/___ -- 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
Re: [asterisk-users] SMS text messaging capabilities
Hello,
Thanks for all of your help. For testing, I am trying to set up a
loopback sms run. However, the sms message does not seem to go
through. Here are the details of the the loopback test:
1. The initial sms send command:
smsq --motx-channel=Zap/g3/2285267 7286657 test
Note that 7285267 is a DID that directs back into the Asterisk system
over the digium Zap device.
2. Here the dialplan for handling the incoming sms call:
exten = 7285267,1,noop(SMS ${EXTEN})
exten = 7285267,2,sms(${EXTEN}|as)
exten = 7285267,3,hangup
I would assume that this dialplan will simply process the incoming
call and save the message in the /var/spool/asterisk/sms/ directory.
However, no incoming message appears in any directory. Furthermore,
no sms log file ever appears in the /var/log/asterisk directory
either.
This is the response I get from the the Asterisk console:
-- Attempting call on Zap/g3/7285267 for application SMS(0) (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
-- Accepting call from '7285660' to '7285267' on channel 0/23, span 3
-- Executing [7285...@pbx:1] NoOp(Zap/85-1, SMS 7285660) in new stack
-- Executing [7285...@pbx:2] SMS(Zap/85-1, 7285660|as) in new stack
-- SMS TX 93 00 6D
[Dec 26 12:23:13] WARNING[2647]: pbx.c:5170 ast_pbx_outgoing_app:
Zap/63-1 already has a call record??
Channel Zap/63-1 was answered.
Launching SMS(0) on Zap/63-1
== Spawn extension (pbx-transfer-mor-redirect, 0772285296, 2) exited
non-zero on 'Zap/85-1'
-- Hungup 'Zap/85-1'
-- Channel 0/1, span 3 got hangup request, cause 16
-- Hungup 'Zap/63-1'
[Dec 26 12:23:23] NOTICE[2647]: pbx_spool.c:351 attempt_thread: Call
completed to Zap/g3/7285267
Thanks for any help,
Elliot
On Sun, Dec 21, 2008 at 1:57 PM, Hans Witvliet h...@a-domani.nl wrote:
On Sat, 2008-12-20 at 19:27 +0200, Elliot Murdock wrote:
Hello!
What kind of sms text messaging capabilities does Asterisk have?
I do not know very much about about SMS technology, but I am looking
for the following features:
1. mobile SIP devices can send and receive SMS messages
2. Asterisk server be able to accept and send SMS messages through PRI
lines and Internet connections.
I noticed that Asterisk has an SMS function, but I am not farmiliar
enough with that technology to make it useful.
Any help with this would be great!
Hi Elliot,
sms-service is included in Asterisk since 1.2.
Just tried it out, (from cli):
smsq --motx-channel=mISDN/1/067364 061368506 testje
=first number is the fixed-sms-provider (here, KPN)
=second number is the target
Just taken rightout from the wiki pages.
I presume one could set the MSN-as well,
hw
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Re: [asterisk-users] SMS text messaging capabilities
Hello,
Sorry, just to avoid confusion, in my last post, the proper smsq command is:
smsq --motx-channel=Zap/g3/7285267 7286657 test
and not to the number 2285267 as stated in the previous post.
Elliot
On Fri, Dec 26, 2008 at 12:26 PM, Elliot Murdock murdo...@gmail.com wrote:
Hello,
Thanks for all of your help. For testing, I am trying to set up a
loopback sms run. However, the sms message does not seem to go
through. Here are the details of the the loopback test:
1. The initial sms send command:
smsq --motx-channel=Zap/g3/2285267 7286657 test
Note that 7285267 is a DID that directs back into the Asterisk system
over the digium Zap device.
2. Here the dialplan for handling the incoming sms call:
exten = 7285267,1,noop(SMS ${EXTEN})
exten = 7285267,2,sms(${EXTEN}|as)
exten = 7285267,3,hangup
I would assume that this dialplan will simply process the incoming
call and save the message in the /var/spool/asterisk/sms/ directory.
However, no incoming message appears in any directory. Furthermore,
no sms log file ever appears in the /var/log/asterisk directory
either.
This is the response I get from the the Asterisk console:
-- Attempting call on Zap/g3/7285267 for application SMS(0) (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
-- Accepting call from '7285660' to '7285267' on channel 0/23, span 3
-- Executing [7285...@pbx:1] NoOp(Zap/85-1, SMS 7285660) in new stack
-- Executing [7285...@pbx:2] SMS(Zap/85-1, 7285660|as) in new stack
-- SMS TX 93 00 6D
[Dec 26 12:23:13] WARNING[2647]: pbx.c:5170 ast_pbx_outgoing_app:
Zap/63-1 already has a call record??
Channel Zap/63-1 was answered.
Launching SMS(0) on Zap/63-1
== Spawn extension (pbx-transfer-mor-redirect, 0772285296, 2) exited
non-zero on 'Zap/85-1'
-- Hungup 'Zap/85-1'
-- Channel 0/1, span 3 got hangup request, cause 16
-- Hungup 'Zap/63-1'
[Dec 26 12:23:23] NOTICE[2647]: pbx_spool.c:351 attempt_thread: Call
completed to Zap/g3/7285267
Thanks for any help,
Elliot
On Sun, Dec 21, 2008 at 1:57 PM, Hans Witvliet h...@a-domani.nl wrote:
On Sat, 2008-12-20 at 19:27 +0200, Elliot Murdock wrote:
Hello!
What kind of sms text messaging capabilities does Asterisk have?
I do not know very much about about SMS technology, but I am looking
for the following features:
1. mobile SIP devices can send and receive SMS messages
2. Asterisk server be able to accept and send SMS messages through PRI
lines and Internet connections.
I noticed that Asterisk has an SMS function, but I am not farmiliar
enough with that technology to make it useful.
Any help with this would be great!
Hi Elliot,
sms-service is included in Asterisk since 1.2.
Just tried it out, (from cli):
smsq --motx-channel=mISDN/1/067364 061368506 testje
=first number is the fixed-sms-provider (here, KPN)
=second number is the target
Just taken rightout from the wiki pages.
I presume one could set the MSN-as well,
hw
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Re: [asterisk-users] Problem: no such extension 'xx' in context 'default'
sip.conf : [general] port=5060 bindaddr=0.0.0.0 put context=default here [10] type=friend secret=1234 host=dynamic context=internal [11] type=friend secret=1234 host=dynamic context=internal extensions.conf [default] exten =2,1,Playback(digits/2) ; exten =2,2,Goto(default,10,1) exten=3,1,Playback(pbx-invalid) exten=3,2,Goto(default,4,1) exten=4,1,Playback(vm-goodbye) exten=4,2,Hangup() Change it to the following: exten =_2,1,Playback(digits/2) ; exten =_2,n,Goto(default,10,1) exten=_3,1,Playback(pbx-invalid) exten=_3,n,Goto(default,4,1) exten=_4,1,Playback(vm-goodbye) exten=_4,n,Hangup() [internal] exten = 10,1,Dial(SIP/10,10) exten =10,2,Background(vm-nobodyavail) exten = 11,1,Dial(SIP/11,5) exten =11,2,Background(vm-nobodyavail) now when I dial 10, I got the following error : no such extension '10' in context 'default' Change it to the following: exten = _10,1,Dial(SIP/10,10) exten =_10,n,Background(vm-nobodyavail) exten = _11,1,Dial(SIP/11,5) exten =_11,n,Background(vm-nobodyavail) The only time I am aware of that you can leave out the prefix underscore is for exten = s and exten = i Hope this helps, Michael ___ -- 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
Re: [asterisk-users] Problem: PART TWO
now when I dial 10, I got the following error : no such extension '10' in context 'default' As anorther important note, your PBX is correct. You should change the line Goto(default,10,1) to Goto(internal,10,1) assuming that's what you want! ___ -- 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
Re: [asterisk-users] Problem: no such extension 'xx' in context 'default'
I was typing so quick I made some slips- (anyway you should get the idea...) Change it to the following: exten =_2,1,Playback(digits/2) ; exten =_2,n,Goto(default,10,1) exten=_3,1,Playback(pbx-invalid) exten=_3,n,Goto(default,4,1) exten=_4,1,Playback(vm-goodbye) exten=_4,n,Hangup() [internal] exten = 10,1,Dial(SIP/10,10) exten =10,2,Background(vm-nobodyavail) exten = 11,1,Dial(SIP/11,5) exten =11,2,Background(vm-nobodyavail) now when I dial 10, I got the following error : no such extension '10' in context 'default' Change it to the following: exten = _10,1,Dial(SIP/10,10) exten =_10,n,Background(vm-nobodyavail) exten = _11,1,Dial(SIP/11,5) exten =_11,n,Background(vm-nobodyavail) The only time I am aware of that you can leave out the prefix underscore is for exten = s and exten = i Hope this helps, Michael ___ -- 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
Re: [asterisk-users] Problem: no such extension 'xx' in context 'default'
Thanks for your quick reply. Now it works. thanks best regards manfred From: mich...@networkstuff.co.nz To: asterisk-users@lists.digium.com Date: Fri, 26 Dec 2008 23:35:34 +1300 Subject: Re: [asterisk-users] Problem: no such extension 'xx' in context 'default' I was typing so quick I made some slips- (anyway you should get the idea...) Change it to the following: exten =_2,1,Playback(digits/2) ; exten =_2,n,Goto(default,10,1) exten=_3,1,Playback(pbx-invalid) exten=_3,n,Goto(default,4,1) exten=_4,1,Playback(vm-goodbye) exten=_4,n,Hangup() [internal] exten = 10,1,Dial(SIP/10,10) exten =10,2,Background(vm-nobodyavail) exten = 11,1,Dial(SIP/11,5) exten =11,2,Background(vm-nobodyavail) now when I dial 10, I got the following error : no such extension '10' in context 'default' Change it to the following: exten = _10,1,Dial(SIP/10,10) exten =_10,n,Background(vm-nobodyavail) exten = _11,1,Dial(SIP/11,5) exten =_11,n,Background(vm-nobodyavail) The only time I am aware of that you can leave out the prefix underscore is for exten = s and exten = i Hope this helps, Michael ___ -- 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 _ Teste jetzt die neue Windows Live Messenger Beta! http://download.live.com/___ -- 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
Re: [asterisk-users] Experiences with grandstream GXW 4024 FXS?
Grandstream GXW 4024 FXS, have audio problems and bad voice quality. - Original Message - From: Mehdi chouikh To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, December 25, 2008 12:56 PM Subject: Re: [asterisk-users] Experiences with grandstream GXW 4024 FXS? I have 4 instalations with with it, the first one from agust, and witout incident, in this time we make only 1 restart. Regards Mehdi Chouikh http://www.voz-ip.info http://www.unitelexperts.com http://www.mitelefonovirtual.com On Wed, Dec 24, 2008 at 6:05 PM, Daniel Hazelbaker dan...@highdesertchurch.com wrote: I use the GXW-4008 and have never had any problems with it. Right now it runs 3 analog phones, but we were using it to link our old NEC phone system to the new Asterisk system, so it was used quite a bit and never once had an issue. Daniel On Dec 24, 2008, at 5:30 AM, Hector Quiroz wrote: HI all, does anyone already implemented the GXW-4024 FXS? Some distributors doesn't recommend it for high volume operations. regards, Hector. ___ -- 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 ___ -- 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 -- Mehdi Chouikh http://www.voz-ip.info http://www.unitelexperts.com http://www.mitelefonovirtual.com -- ___ -- 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___ -- 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
Re: [asterisk-users] Problem: no such extension 'xx' in context 'default'
Michael wrote: Change it to the following: exten = _10,1,Dial(SIP/10,10) exten =_10,n,Background(vm-nobodyavail) exten = _11,1,Dial(SIP/11,5) exten =_11,n,Background(vm-nobodyavail) The only time I am aware of that you can leave out the prefix underscore is for exten = s and exten = i No, you can leave out the leading underscore when you are using explicit extension numbers. (such as those above) As soon as you introduce /any/ pattern matching characters, you /must/ include the leading underscore. ___ -- 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
[asterisk-users] Meetme - play the name
Hi, I have a requirement, whenever a user comes into the conference, it has to announce the user name to all the person who are all available in the conference. I have used Meetme(,di) where i is to announce the user leave/join with review. I user used I also, which is to announce the user leave/join with out review. In both the above cases, it is prompting the user to say their name, but what i want is, if it gets the name one time, thats all, it should just play that name whenever the call comes from the same callerid. Is it possible to achieve this feature by some way? Hope somebody would have the same requirement, kindly help to achieve to do the same. thanks and regards Sasikala. Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/___ -- 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
[asterisk-users] Wich gateway is much better?
Hello everybody, I have a doubt If I want to send every call from a server asterisk to a gateway to a line PSTN, in the gateway what type of port I need FXO o FXS? I need to know wich gateway to buy, with port FXS or FXO? regards, Abel ___ -- 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
Re: [asterisk-users] Wich gateway is much better?
On Sat, 27 Dec 2008 18:42:12 Abel Monzon wrote: Hello everybody, I have a doubt If I want to send every call from a server asterisk to a gateway to a line PSTN, in the gateway what type of port I need FXO o FXS? I need to know wich gateway to buy, with port FXS or FXO? regards, Abel FXO = Goes to PSTN FXS = Goes to telephone Michael ___ -- 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
Re: [asterisk-users] Meetme - play the name
On Sat, Dec 27, 2008 at 10:07 AM, sasikala kala sasi_jeyalaks...@yahoo.comwrote: Hi, I have a requirement, whenever a user comes into the conference, it has to announce the user name to all the person who are all available in the conference. I have used Meetme(,di) where i is to announce the user leave/join with review. I user used I also, which is to announce the user leave/join with out review. In both the above cases, it is prompting the user to say their name, but what i want is, if it gets the name one time, thats all, it should just play that name whenever the call comes from the same callerid. Is it possible to achieve this feature by some way? Hope somebody would have the same requirement, kindly help to achieve to do the same. thanks and regards Sasikala. On a quick glance at meetme application options, I don't see any such direct facility that would let you do what you want. But you can try out 'b' option of meetme to run a AGI+AMI mixed script in background to detect entry of a new member and then play out their name. AGI and AMI are very powerful you can easily figure out some solution with them -- Thanks Regards, Godson Gera Asterisk Consultant Hyderabadhttp://godson.in/voip-asterisk-consultant-hyderabad-india ___ -- 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
