[asterisk-users] User registration failure bug ?
Hi list,
I have a user, referenced by his IMSI (IMSI208300618462231), who is
assigned to extension 2111 in /etc/asterisk/extensions.conf and
sip.conf (see below).
From time to time, registration of this user fails (see below), but I
do not know why. Anybody has a clue what could be wrong ? Is this a
bug ?
[I rebooted asterisk, and now it works.]
Regards
Axelle.
Logs of failed registration:
sip show users
Username Secret Accountcode
Def.Context ACL NAT
IMSI208011234567890 sip-local
No RFC3581
IMSI208302141472352
sip-external No RFC3581
IMSI208304424439206
sip-external No RFC3581
[Apr 8 15:01:01] NOTICE[20626]: chan_sip.c:15642
handle_request_register: Registration from 'IMSI208300618462231
sip:IMSI208300618462231@127.0.0.1' failed for '127.0.0.1' - No
matching peer found
sip show user IMSI208300618462231
User IMSI208300618462231 not found.
My configuration in extensions.conf:
[IMSI208300618462231]
callerid=2111
canreinvite=no
type=friend
context=sip-external
allow=gsm
host=dynamic
sip.conf:
exten = 2111,1,Macro(dialSIP,IMSI208300618462231)
where dialSIP is a macro:
[macro-dialSIP]
exten = s,1,Dial(SIP/${ARG1})
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Busy(3000)
exten = s-CONGESTION,1,Congestion(3000)
exten = s-CHANUNAVAIL,1,playback(ss-noservice)
exten = s-CANCEL,2,Hangup
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Re: [asterisk-users] Registration failed though configured.
Hi, Do these IMSI names / numbers match what your phone is trying to register as? Are there actual at the end of the numbers, or are you attempting to obfuscate? yes xxx are numbers (not real letters x), it's just 'obfuscation' and anyway it's easier to recognize them by the first few digits. and yes, they match the phone. Show us the actual logs and the actual sip.conf well, there isn't more apart from the macro... And I don't understand why it would work for one phone and not the other one when the configuration is the same. Is it possible someway to clear the HLR, databases etc and be sure to restart in a clean state? -- Axelle -- _ -- 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
[asterisk-users] Registration failed though configured.
Hi list, Currently, one of my phones registers fine, and the other does not, though for me they have the same config... Can somebody help debug/understand why? The logs in asterisk say: [Feb 24 13:48:09] NOTICE[20626]: chan_sip.c:15642 handle_request_register: Registration from 'IMSI208300618462231 sip:IMSI20830061@127.0.0.1' failed for '127.0.0.1' - No matching peer found Thanks in /etc/asterisk/extensions.conf: exten = 2102,1,Macro(dialSIP,IMSI2081) ; this one registers ok exten = 2111,1,Macro(dialSIP,IMSI20830061) ; fails In sip.conf: [IMSI2081] ; callerid=2102 canreinvite=no type=friend context=sip-external allow=gsm host=dynamic [IMSI20830061] callerid=2111 canreinvite=no type=friend context=sip-external allow=gsm host=dynamic -- _ -- 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] Assigning an extension to a roaming phone
So you have an IP network, with SIP agents (cell phones ?), some of those are manually setup in you sip.conf file, but you want to allow unknown cell phones users to self register in your system ? Yes, exactly. Someone enter your network, dial 3001@your ipbx and get/set a temporary internal number. Then other phone can dial his ? Yes, right. I don't think it's possible, although ... What you need is to mimic the SIP registration process, by fetching the following informations from during the setup call: * IP of the phone * UDP/TCP Port of the SIP process * Some SIP user ID Then you store thoses in your DB in the form SIP/user@IP:port and then you could be able to Dial this string, (if the phone is ok to be dialed by an unknown party this way) Mmm. yes, but I don't have a clue how I could do that. Perhaps also if I were able to retrieve the IMSI of the roaming user I might be able to work out something. But I don't know how to get it... -- Axelle -- _ -- 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] Assigning an extension to a roaming phone
Hi Danny, That's a nice log I'll try and do the same with a higher verbosity level on my side too. Just to make sure - who called 3001? the roaming phone that had no extension yet? -- Executing [3001@default:1] Verbose(SIP/sipuser-006f, Create roaming extension) in new stack - when you called 4144 (from another phone), it triggers 144 - which I understand - but did that 144 actually have the roaming phone ring? -- Executing [4144@default:2] Set(SIP/sipuser-0070, ROAMEXT=144) in new stack Thanks Axelle -- _ -- 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] Assigning an extension to a roaming phone
exten = 3001,n,playback(vm-youhave)
I do have the file in /usr/share/asterisk/sounds:
-rw-r--r-- 1 root root 1452 2008-03-06 00:39 vm-youhave.gsm
but still it does not play it ?!
The goodbye at the end does play correctly.
** vm-goodbye is in /usr/share/asterisk/sounds?
Yes it is.
$ ls -al /usr/share/asterisk/sounds/vm-goodb*
-rw-r--r-- 1 root root 1683 2008-03-06 00:39
/usr/share/asterisk/sounds/vm-goodbye.gsm
Part two
exten = _4XXX,1,Set(ROAM=${DB(roam/ext)})
exten = _4XXX,n,Dial(SIP/${ROAM},30,,mKkTt)
line 1 user dials 4001 and gets ${ROAM} set from ASTDB
line 2 attempts to dial SIP extension based on ${ROAM} value.
I dialed 3001, then 001. It does say 001 back.
But then 4001 does not work.
[Feb 21 17:53:06] WARNING[26195]: chan_sip.c:2921 create_addr: No such host:
001
[Feb 21 17:53:06] WARNING[26195]: app_dial.c:1202 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
-- Axelle
I doubt you have an extension 001 in your list (the number 4001 is trying to
dial).
Well, the 001 is supposed to be created by this line, isn't it?
exten = 3001,n,Set(DB(roam/ext)=${digito})
But obviously, yes, it is not working :(
Is the ${ROAM} trying to reach an in-house extension or an outside
number?
an in-house extension.
-- Axelle
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Re: [asterisk-users] Assigning an extension to a roaming phone
[roaming-ext]
;Create a new roaming extension
exten = 3001,1(readop),Verbose(Create roaming extension)
exten = 3001,n,Read(digito,beep,3)
exten = 3001,n,Playback(you-entered)
exten = 3001,n,SayDigits(${digito})
exten = 3001,n,Verbose(Setting roaming extension 4${digito} to call
${CALLERID(num)})
exten = 3001,n,Set(DB(roam/${digito})=${CALLERID(num)})
exten = 3001,n,Playback(vm-goodbye)
exten = 3001,n,Hangup()
Good idea the Verbose commands, at least I see a bit better what is
happening. I should have thought about that one. Thanks.
But I don't understand the CALLERID part: the roaming user is unknown
on my network, so how could he have a correct CALLERID?
;Dial a roaming extension
exten = _4XXX,1,Verbose(Calling roaming extension ${EXTEN})
exten = _4XXX,n,Set(ROAMEXT=${DB(roam/${EXTEN:1})})
exten = _4XXX,n,Dial(SIP/${ROAMEXT},30)
I tried it and I get the following logs:
[Feb 22 11:57:44] NOTICE[20626]: chan_sip.c:15642
handle_request_register: Registration from 'IMSI20830'
sip:IMSI20830@127.0.0.1' failed for '127.0.0.1' - No matching
peer found
= this line appears when the roaming user comes in
Create roaming extension
Setting roaming extension 4001 to call 2103
= those two lines occur when the roaming user dials 3001. Why is it
returning a callerid 2103?? 2103 corresponds to another registered
IMSI !!
Calling roaming extension 4001
[Feb 22 12:05:07] WARNING[27577]: chan_sip.c:2921 create_addr: No such
host: 2103
[Feb 22 12:05:07] WARNING[27577]: app_dial.c:1202 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
= this is displayed when the roaming user dials 4001. It's quite
normal it can't route to 2103 because that phone is off.
Then you need to include the [roaming-ext] context in whatever context your
phones dial from. The basic idea behind this is that you need to store the
extension where your roamer is currently sitting in your DB, which you were
doing. By adding the ${CALLERID(num)} to the database, you give it an idea
of where the calls should go.
Not sure to understand.
The goal here is to assign an extension to the roaming user. He calls 3001.
Where 'the calls should go' is to the roaming user.
Now, this means your ${CALLERID(num)}
variable needs to match your SIP endpoint's name, of course, but if these
don't currently match, I'm pretty sure there is a variable you can use to
achieve the same effect.
-- Axelle
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Re: [asterisk-users] Assigning an extension to a roaming phone
H Danny,
Axelle, please post the CLI output from the 3001 call and I'll put up a
dialplan that should work for you.
So this is the output I get:
Connected to Asterisk 1.4.21.2~dfsg-3+lenny1 currently running on
openbts (pid = 20597)
[Feb 22 15:18:02] NOTICE[20626]: chan_sip.c:15642
handle_request_register: Registration from
'IMSI20830061sip:IMSI20830061@127.0.0.1' failed for
'127.0.0.1' - No matching peer found
Create roaming extension
Caller IMSI is
Setting roaming extension 4001
Calling roaming extension 4001
[Feb 22 15:22:42] WARNING[9512]: chan_sip.c:2921 create_addr: No such host: 4001
[Feb 22 15:22:42] WARNING[9512]: app_dial.c:1202 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
openbts*CLI database show
/SIP/Registry/IMSI20810 :
127.0.0.1:5062:3600:IMSI20810:sip:IMSI20810@127.0.0.1:5062
/SIP/Registry/IMSI20830061 :
127.0.0.1:5062:3600:IMSI20830061:sip:IMSI20830061@127.0.0.1:5062
/SIP/Registry/IMSI2083044xxx :
127.0.0.1:5062:3600:IMSI2083044:sip:IMSI2083044@127.0.0.1:5062
/roam/001 : 4001
/roam/002 : 2103
/roam/003 : 4003
/roam/007 : 4007
/roam/ext : 001
openbts*CLI
My current extensions.conf is
[globals]
; This is the extensions file used in the Burning Man 2008
; site test, with private information removed.
; Jump to the end for handset examples.
[macro-dialSIP]
exten = s,1,Dial(SIP/${ARG1})
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Busy(3000)
exten = s-CONGESTION,1,Congestion(3000)
exten = s-CHANUNAVAIL,1,playback(ss-noservice)
exten = s-CANCEL,2,Hangup
[from-trunk]
; route incoming calls from the PSTN
[sip-external]
include = sip-local
; roaming users
;Create a new roaming extension
exten = 3001,1(readop),Verbose(Create roaming extension)
exten = 3001,n,Verbose(Caller IMSI is ${IMSI})
exten = 3001,n,Read(digito,beep,3)
exten = 3001,n,Playback(vm-goodbye)
exten = 3001,n,SayDigits(${digito})
exten = 3001,n,Verbose(Setting roaming extension 4${digito})
exten = 3001,n,Set(DB(roam/${digito})=4${digito})
exten = 3001,n,Playback(vm-goodbye)
exten = 3001,n,Hangup()
;Dial a roaming extension
exten = _4XXX,1,Verbose(Calling roaming extension ${EXTEN})
exten = _4XXX,n,Set(ROAMEXT=${DB(roam/${EXTEN:1})})
exten = _4XXX,n,Dial(SIP/${ROAMEXT},30)
; outgoing trunk access
; NANP
[sip-local]
; removing full IMSI value for the cut and paste in the list
exten = 2102,1,Macro(dialSIP,IMSI20810)
exten = 2103,1,Macro(dialSIP,IMSI2083044)
; Note 2111 is commented:
;exten = 2111,1,Macro(dialSIP,IMSI20830061)
exten = 6123,1,SayNumber(${EXTEN})
My sip.conf is:
[general]
bindaddr=0.0.0.0
bindport=5060
; Comment these out if no backhaul is available.
; This is a GSM handset entry.
; You need one for each SIM.
; The IMSI is a 15-digit code in the SIM.
; You can see it in the Control log whenever a phone tries to register.
[IMSI20810]
callerid=2102
canreinvite=no
type=friend
context=sip-external
allow=gsm
host=dynamic
[IMSI2083044]
callerid=2103
canreinvite=no
type=friend
context=sip-external
allow=gsm
host=dynamic
;[IMSI20830061]
;callerid=2111
;canreinvite=no
;type=friend
;context=sip-external
;allow=gsm
;host=dynamic
-- Axelle
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Re: [asterisk-users] Assigning an extension to a roaming phone
Axelle, please post the CLI output from the 3001 call and I'll put up a dialplan that should work for you. snip Not what I asked for, but here's what I can tell you. Oh I'm sorry but then what are you asking for? I thought it was the console messages on Asterisk. From what you posted, you can dial and outside number and from in-house you can dial 2102 or 2103. The way the dialplan works is that you set up specific numbers that will be valid like you have done with 2102, 2103 and 3001 or a range of numbers that will be valid like 4000-4999. For the 4XXX magic number snippet to ever work correctly, it has to dial an outside number or a pre-defined in-house extension. From what you posted, if you dial 4002, the call should properly connect to 2103. Yes, indeed, but that's not what I want it to do. 2103 does not correspond to anyone. Yeah, by the way, just to make that clear: the roaming phone does not have *any phone number*. I need the dialplan to assign one. Re-routing to another number won't work, as there is no other number... Thanks Axelle -- _ -- 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] Assigning an extension to a roaming phone
Hi Danny,
Thanks again for your help !
Exten = 4001,1,Dial(DAHDI/g1/5551212)
Or
Exten = 4001,1,Dial(SIP/5551...@myprovider.com)
It looks like I'd be using Dial(SIP/...) as for other numbers I have a
macro such as this:
[macro-dialSIP]
exten = s,1,Dial(SIP/${ARG1})
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-CANCEL,1,Hangup
etc
line 2 plays enter roaming number (you have to record it)
well, I'm trying to use a pre-recorded sound.
But it does not play it.
exten = 3001,n,playback(vm-youhave)
I do have the file in /usr/share/asterisk/sounds:
-rw-r--r-- 1 root root 1452 2008-03-06 00:39 vm-youhave.gsm
but still it does not play it ?!
The goodbye at the end does play correctly.
Part two
exten = _4XXX,1,Set(ROAM=${DB(roam/ext)})
exten = _4XXX,n,Dial(SIP/${ROAM},30,,mKkTt)
line 1 user dials 4001 and gets ${ROAM} set from ASTDB
line 2 attempts to dial SIP extension based on ${ROAM} value.
I dialed 3001, then 001. It does say 001 back.
But then 4001 does not work.
[Feb 21 17:53:06] WARNING[26195]: chan_sip.c:2921 create_addr: No such host: 001
[Feb 21 17:53:06] WARNING[26195]: app_dial.c:1202 dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to
destination)
-- Axelle
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[asterisk-users] Assigning an extension to a roaming phone
Hi,
I'm trying to automatically have the dialplan assign an extension to a
roaming phone on my network.
I tried the following without success:
exten = 3001,1(readop),BackGround(beep)
exten = 3001,n,Read(digito,vm-youhave,3)
exten = 3001,n,SayDigits(${digito})
exten = 3001,n,Set(ROAM=${digito})
exten = 3001,n,Set(DB(roam/ext)=${digito})
exten = 3001,n,playback(vm-goodbye)
exten = 3001,n,hangup
exten = _4XXX,1,Set(ROAM=${DB(roam/ext)})
exten = _4XXX,n,dial(SIP/${ROAM})
The idea was that the roaming phone first dials 3001, sets a 3 digits
extension (eg 123) and then I supposed that 4123 would work. But it
does not.
I am unsure about the 2 Set lines.
Can anyone help?
Regards
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Re: [asterisk-users] Assigning an extension to a roaming phone
Hi,
I'm thinking that _4XXX is an over-complication. _4XXX means you could
dial any number from 4000 through 4999 inclusive and get the extension at
SIP/${ROAM}.
Well it's kind of what I want.
I have a roaming phone that comes in. He dials 3001, sets his
extension to 123, so that he is assigned 4123.
I have another roamding phone that comes in. Dials 3001, sets his
extension to 124. He is assigned 4124.
Or at least that's how I understand it.
In reality, what I only need is all roaming phones to get assigned an
extension (within a given range) and to have a way to find their
extension number.
Roaming phone 1 comes in. Get assigned (automatically) 4001.
Roaming phone 2 comes in. Get assigned (automatically) 4002.
Roaming phone 3 comes in. Get assigned (automatically) 4003.
etc
I'd change the line 2
- exten = _4XXX,n,Dial(SIP/${ROAM},30,,mKkTt)
looks pretty similar to the previous line - apart from that mKkTt.
What is that for?
What's wrong with the previous line.
Or
- exten = 4123,1,Dial(SIP/${ROAM},30,,mKkTt)
That would only match case where roaming user wants to be assigned
4123, but it would not work for - say - 4124.
Thanks.
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Re: [asterisk-users] How to list used extensions + assign extension toa roaming phone
Hi Danny,
For question 1, I think sip show peers is what you want.
Yes, indeed. Thanks.
Though, there's something strange with it, but probably related to
question 2 below.
For question 2,
here are two ways to do it.
I tried both ways but couldn't get it working. In both cases, this is
what happened:
I have phones (non-roaming) to which I have assigned numbers such as
2102, 2103 etc.
The roaming phone can call the other phones (for eg 2103) without problem.
But strange, when he calls 2103, 2103 sees an incoming call of 2111,
which is the last of the non-roaming phone numbers I configured. And
on asterisk command line,
sip show peers
shows as if 2103 and 2111 were online
and sip show channels
shows as if 2103 and 2111 were in the middle of their call !
Though, I checked it, 2111 is offline. The only two phones online are
2103 and the roaming phone.
Looks like a bug in my config, huh ?
Now, the other way round, 2103 can't call the roaming phone (which is
what I would have liked). I tried 2103 call 2111, but that doesn't
work (and confirms 2111 is offline). I tried 2103 calls 3000 or 3001,
no success.
Can somebody help debug this?
Here is my config:
[sip-local]
exten = 2103,1,Macro(dialSIP,IMSIA)
exten = 2104,1,Macro(dialSIP,IMSIB)
exten = 2105,1,Macro(dialSIP,IMSIC)
exten = 2110,1,Macro(dialSIP,IMSID)
exten = 2111,1,Macro(dialSIP,IMSIE)
; roaming users
; I tried both solutions:
;exten = 3xxx,1,dial(SIP/foo)
exten = 3001,1(readop),BackGround(beep)
exten = 3001,n,Read(digito,assignroam,3)
exten = 3001,n,SayDigits(${digito})
exten = 3001,n,Set(ROAM=${digito})
exten = 3001,n,Set(DB(roam/ext)=${digito})
exten = 3001,n,playback(vm-goodbye)
exten = 3001,n,hangup
exten = 4xxx,1,Set(ROAM=${DB(roam/ext)})
exten = 4xxx,n,dial(SIP/${ROAM})
Regards
Axelle.
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Re: [asterisk-users] setting up callerid
Hi Dave, context=openbts callerid=473520 I see you are using OpenBTS. To my understanding, OpenBTS does not support caller ID, so I don't think it can work. But as I have the same issue as you, I'd be glad to be wrong ! :D Let me know. Disregard my answer. I just tested the callerid on my OpenBTS and it worked. So the problem you encounter must be elsewhere. Regards Axelle -- _ -- 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
[asterisk-users] How to list used extensions + assign extension to a roaming phone
Hi list, I have searched through asterisk command lines but haven't found how to do this: - can I list the phones (callerid or IMSIs?) currently registered ? If I do dialplan show that lists the configuration I loaded, e.g [ Context 'sip-local' created by 'pbx_config' ] '2102' = 1. Macro(dialSIP|IMSI1) [pbx_config] '2103' = 1. Macro(dialSIP|IMSI2) [pbx_config] '2104' = 1. Macro(dialSIP|IMSI3) [pbx_config] but it does not tell me who is actually registered or using the network, maybe only 2102. - is it possible to assign a given number/range of numbers (extension) to a phone which roams into my network (open registration)? Thanks Axelle -- _ -- 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] setting up callerid
Hi Dave,
On Thu, Dec 16, 2010 at 1:52 PM, dave george dgeo...@teletoneinc.com wrote:
Tried the following but no luck:
exten = _53.,1,Set(CALLERID(num)=473520)
exten = _53.,n,Dial(SIP/${ext...@ss74)
I am still passing IMSI310410381554227 as the CALLERID.
My peer is setup as follows:
[IMSI310410381554227]
canreinvite=no
type=peer
context=openbts
callerid=473520
I see you are using OpenBTS. To my understanding, OpenBTS does not
support caller ID, so I don't think it can work.
But as I have the same issue as you, I'd be glad to be wrong ! :D Let me know.
Regards
Axelle
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