Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
Vic Cross wrote: On Sat, 7 Feb 2004, John Fraizer wrote: Here are the configs: ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 66.35.64.38 ; Address to bind to context = default ; Default for incoming calls srvlookup = yes ; Enable SRV lookups on outbound calls [100] type=friend username=100 secret=secret host=dynamic fromuser=100 mailbox=100 context=allaccess canreinvite=yes dtmfmode=rfc2833 nat=yes [228] type=friend username=228 secret=secret host=dynamic fromuser=228 mailbox=100 context=allaccess canreinvite=yes dtmfmode=rfc2833 nat=yes [] type=friend username= secret=secret host=dynamic fromuser= context=allaccess canreinvite=yes dtmfmode=rfc2833 nat=yes Remove "fromuser=" from your SIP statements. This overrides the caller-id data received with whatever is stated in fromuser -- Asterisk is doing exactly what you told it to. ;-) To explain a bit more: Fromuser= and fromdomain= is used to specify the caller when calling this device. This is used when we're calling a SIP proxy that requires a specific fromuser/domain in addition to an authentication. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
On Sat, 7 Feb 2004, John Fraizer wrote: > Here are the configs: > > ; > ; SIP Configuration for Asterisk > ; > [general] > port = 5060 ; Port to bind to > bindaddr = 66.35.64.38 ; Address to bind to > context = default ; Default for incoming calls > srvlookup = yes ; Enable SRV lookups on outbound calls > > > [100] > type=friend > username=100 > secret=secret > host=dynamic > fromuser=100 > mailbox=100 > context=allaccess > canreinvite=yes > dtmfmode=rfc2833 > nat=yes > > [228] > type=friend > username=228 > secret=secret > host=dynamic > fromuser=228 > mailbox=100 > context=allaccess > canreinvite=yes > dtmfmode=rfc2833 > nat=yes > > [] > type=friend > username= > secret=secret > host=dynamic > fromuser= > context=allaccess > canreinvite=yes > dtmfmode=rfc2833 > nat=yes Remove "fromuser=" from your SIP statements. This overrides the caller-id data received with whatever is stated in fromuser -- Asterisk is doing exactly what you told it to. ;-) Hoo-roo, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
> OK. I don't know what the deal is. Works fine on your server. Doesn't on > mine. > > That is so strange. my version string is: CVS-01/31/04-04:24:34 Also, I noticed that your sip.conf entries are a bit different than mine. I am curious if canreinvite=no would change your situation. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
Robert Hajime Lanning wrote: I can do this, hold on. OK. I don't know what the deal is. Works fine on your server. Doesn't on mine. That is so strange. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
Olle E. Johansson wrote:
Would like to see a SIP debug
* The invite from the caller phone to Asterisk
* The invite from Asterisk to the called phone
As well as the configs (extensions.conf and sip.conf)
Can't reproduce in my servers.
/O
OK. Here is a call from extension 100 to extension 228.
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 24.33.239.118:5060;branch=z9hG4bK149f400a
From: "John Fraizer 100"
;tag=000bbe40419b00532a4215e9-779f0059
To:
Call-ID: [EMAIL PROTECTED]
Date: Sat, 07 Feb 2004 22:57:46 GMT
CSeq: 101 INVITE
User-Agent: CSCO/6
Contact:
Expires: 180
Content-Type: application/sdp
Content-Length: 249
Accept: application/sdp
v=0
o=Cisco-SIPUA 21234 22236 IN IP4 24.33.239.118
s=SIP Call
c=IN IP4 24.33.239.118
t=0 0
m=audio 18846 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
13 headers, 11 lines
Using latest request as basis request
Sending to 24.33.239.118 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 524302, them - 268/0, combined - 12
Non-codec capabilities: us - 1, them - 1, combined - 1
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
24.33.239.118:5060;branch=z9hG4bK149f400a;received=24.33.239.118
From: "John Fraizer 100"
;tag=000bbe40419b00532a4215e9-779f0059
To: ;tag=as0638308b
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="4844d22f"
Content-Length: 0
to 24.33.239.118:5060
Border2*CLI>
Sip read:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK3579a8a9
From: "John Fraizer 100"
;tag=000bbe40419b00532a4215e9-779f0059
To: ;tag=as0638308b
Call-ID: [EMAIL PROTECTED]
Date: Sat, 07 Feb 2004 22:57:46 GMT
CSeq: 101 ACK
Content-Length: 0
8 headers, 0 lines
Border2*CLI>
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 24.33.239.118:5060;branch=z9hG4bK3f142fb1
From: "John Fraizer 100"
;tag=000bbe40419b00532a4215e9-779f0059
To:
Call-ID: [EMAIL PROTECTED]
Date: Sat, 07 Feb 2004 22:57:46 GMT
CSeq: 102 INVITE
User-Agent: CSCO/6
Contact:
Proxy-Authorization: Digest
username="100",realm="asterisk",uri="sip:66.35.64.38",response="9d7ae43306bc23bb256068b8f4044017",nonce="4844d22f",algorithm=md5
Expires: 180
Content-Type: application/sdp
Content-Length: 249
v=0
o=Cisco-SIPUA 21234 22236 IN IP4 24.33.239.118
s=SIP Call
c=IN IP4 24.33.239.118
t=0 0
m=audio 18846 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
13 headers, 11 lines
Using latest request as basis request
Sending to 24.33.239.118 : 5060 (NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 524302, them - 268/0, combined - 12
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 228 in allaccess
list_route: hop:
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
24.33.239.118:5060;branch=z9hG4bK3f142fb1;received=24.33.239.118
From: "John Fraizer 100"
;tag=000bbe40419b00532a4215e9-779f0059
To: ;tag=as4cba15e7
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
to 24.33.239.118:5060
*** THIS IS WHERE IT STARTS BREAKING ***
-- Executing Dial("SIP/100-9284", "SIP/228|20") in new stack
We're at 66.35.64.38 port 10990
Answering with capability 2
Answering with capability 4
Answering with capability 8
Answering with non-codec capability 1
12 headers, 11 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK1aed7dc4
From: "John Fraizer 100" ;tag=as7e10d688
To:
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 07 Feb 2004 22:57:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 234
v=0
o=root 12840 12840 IN IP4 66.35.64.38
s=session
c=IN IP4 66.35.64.38
t=0 0
m=audio 10990 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
(NAT) to 24.33.239.118:5060
-- Called 228
Border2*CLI>
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK1aed7dc4
From: "John Fraizer 100" ;tag=as7e10d688
To:
Call-ID: [EMAIL PROTECTED]
Date: Sat, 07 Feb 2004 22:57:47 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact:
Content-Length: 0
10 headers, 0 lines
Border2*CLI>
Sip re
Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
Would like to see a SIP debug * The invite from the caller phone to Asterisk * The invite from Asterisk to the called phone As well as the configs (extensions.conf and sip.conf) Can't reproduce in my servers. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
Robert Hajime Lanning wrote:
That is real interesting. It seems to work just fine for me. Though, I am
running straigh out of CVS, but older than 0.7.2 release. My SIP phones
(Grandstream) see CallerID just fine and my co-worker's SIP phones (Cisco 7960)
work also.
Can you send your extensions.conf? It has to be something in there.
The portions that have anything to do with the two extensions I'm testing
with are:
[general]
static=yes
writeprotect=yes
[default]
exten => john,1,Dial(SIP/100,20)
[voicemail]
exten => 8500,1,Ringing
exten => 8500,2,Wait,1
exten => 8500,3,VoicemailMain2(${CALLERIDNUM})
exten => 8500,4,Hangup
exten => 8501,1,Ringing
exten => 8501,2,Wait,1
exten => 8501,3,VoicemailMain2
exten => 8501,4,Hangup
exten => 8502,1,Ringing
exten => 8502,2,Wait,1
exten => 8502,3,Voicemail2(u${CALLERIDNUM})
exten => 8502,4,Hangup
[extensions]
exten => 100,1,Dial(SIP/100,20)
exten => 100,2,Voicemail2(u100)
exten => 100,3,Hangup
exten => 100,102,Voicemail2(b100)
exten => ,1,Dial(SIP/,20)
exten => ,2,Hangup
[allaccess]
include => voicemail
include => extensions
Thanks,
John
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Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
> If your asterisk server does not do this, please do me the favor of setting > up two "test" extensions for me so I can try to figure out what is wrong > here. You can lock me in a context where I can only call from one "test > extension" to another. I just need to be able to verify what is going on so > I can either get it corrected in my config (I don't think I have anything > wrong) or get it acknowledged as a bug in Asterisk. I can do this, hold on. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
I just set up an X-Lite client on extension "" and it shows the same behavior. Further information: When I call the X-Lite extension "" from my Cisco 7960 (extension 100), I get the following in the "recent calls" list: Name: Test SIPURL: [EMAIL PROTECTED] ProxyID: ENTERZONE The sip.conf entry for extension 100 is: [100] callerid=Test <100> type=friend username=100 secret=secret host=dynamic fromuser=100 mailbox=100 context=allaccess canreinvite=yes dtmfmode=rfc2833 nat=yes Asterisk is sending the "name" portion of the callerID properly but, the "number" portion is obviously wrong as you can see from the SIPURL that is saved by X-Lite in the "recent received calls" list. As I have been stating, it is sending the CALLED number and not the "calling" number. This is NOT the proper behavior and as a result, it is hosing caller ID. I could really not care any less about what "name" shows up. The sipurl that called has to be right though, otherwise callerid is worthless. It doesn't do any good to look at your "missed calls" list and have every one of them show YOUR phone number. If your asterisk server does not do this, please do me the favor of setting up two "test" extensions for me so I can try to figure out what is wrong here. You can lock me in a context where I can only call from one "test extension" to another. I just need to be able to verify what is going on so I can either get it corrected in my config (I don't think I have anything wrong) or get it acknowledged as a bug in Asterisk. Thanks, John John Fraizer wrote: Robert Hajime Lanning wrote: OK. I upgraded to 0.7.2 but and also set a "callerid=" entry in sip.conf. The behavior is the same. Caller-ID is sent as "Name of Calling Party" instead of "Name of Calling Party" like it should be. You are not setting the caller ID properly... callerid = "string portion" If you want no string portion, then: callerid = "" Um, yes I am setting the caller ID right. Asterisk isn't sending the invite message properly. [100] callerid= "test name" <1234> type=friend username=100 secret=secret host=dynamic fromuser=100 mailbox=100 context=allaccess canreinvite=yes dtmfmode=rfc2833 nat=yes The "test name" part gets sent but, like I said, if extension 100 calls extension 228, the phone at 228 sees the caller-ID as "test name" <228>. This happens with Asterisk 0.5 and Asterisk 0.7.2 both. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
> Um, yes I am setting the caller ID right. Asterisk isn't sending the invite > message properly. > > [100] > callerid= "test name" <1234> > type=friend > username=100 > secret=secret > host=dynamic > fromuser=100 > mailbox=100 > context=allaccess > canreinvite=yes > dtmfmode=rfc2833 > nat=yes > > The "test name" part gets sent but, like I said, if extension 100 calls > extension 228, the phone at 228 sees the caller-ID as "test name" > <228>. > > This happens with Asterisk 0.5 and Asterisk 0.7.2 both. That is real interesting. It seems to work just fine for me. Though, I am running straigh out of CVS, but older than 0.7.2 release. My SIP phones (Grandstream) see CallerID just fine and my co-worker's SIP phones (Cisco 7960) work also. Can you send your extensions.conf? It has to be something in there. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
Robert Hajime Lanning wrote: OK. I upgraded to 0.7.2 but and also set a "callerid=" entry in sip.conf. The behavior is the same. Caller-ID is sent as "Name of Calling Party" instead of "Name of Calling Party" like it should be. You are not setting the caller ID properly... callerid = "string portion" If you want no string portion, then: callerid = "" Um, yes I am setting the caller ID right. Asterisk isn't sending the invite message properly. [100] callerid= "test name" <1234> type=friend username=100 secret=secret host=dynamic fromuser=100 mailbox=100 context=allaccess canreinvite=yes dtmfmode=rfc2833 nat=yes The "test name" part gets sent but, like I said, if extension 100 calls extension 228, the phone at 228 sees the caller-ID as "test name" <228>. This happens with Asterisk 0.5 and Asterisk 0.7.2 both. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
> >> >> OK. I upgraded to 0.7.2 but and also set a "callerid=" entry in sip.conf. >> The behavior is the same. >> >> Caller-ID is sent as "Name of Calling Party" >> instead of "Name of Calling Party" like it should >> be. > > You are not setting the caller ID properly... > > callerid = "string portion" > > If you want no string portion, then: > > callerid = "" Also, it is the same syntax for the SetCallerID() application. The way you had it: callerid = 200 Sets the string portion to "200" and leaves the number portion null. The null number portion is what is causing you trouble. -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
> > OK. I upgraded to 0.7.2 but and also set a "callerid=" entry in sip.conf. > The behavior is the same. > > Caller-ID is sent as "Name of Calling Party" > instead of "Name of Calling Party" like it should > be. You are not setting the caller ID properly... callerid = "string portion" If you want no string portion, then: callerid = "" -- END OF LINE -MCP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
OK. I upgraded to 0.7.2 but and also set a "callerid=" entry in sip.conf.
The behavior is the same.
Caller-ID is sent as "Name of Calling Party"
instead of "Name of Calling Party" like it should be.
If you look at the sip debug of a call between extenstion 228 and
extension 100, you can see what is causing the problem:
-- Executing Dial("SIP/228-76d1", "SIP/100|20") in new stack
We're at 66.35.64.38 port 13694
Answering with preferred capability 4
Answering with preferred capability 8
Answering with preferred capability 256
Answering with non-codec capability 1
12 headers, 11 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK025b4e85
From: "John Fraizer" ;tag=as2e305230
To:
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 07 Feb 2004 19:07:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 237
v=0
o=root 10168 10168 IN IP4 66.35.64.38
s=session
c=IN IP4 66.35.64.38
t=0 0
m=audio 13694 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
(NAT) to 24.33.239.118:5060
-- Called 100
It is this part that is causing it to get the wrong caller ID number:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK025b4e85
From: "John Fraizer" ;tag=as2e305230
To:
Notice that we're inviting 100@ while claiming that the call is COMING from
[EMAIL PROTECTED] It puts the right caller ID *name* in the invite but, the "sip:100"
in the from field is flat out wrong. It should be "sip:228".
Surely I am not the only one to notice that this is broken.
John
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Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
Hi William,
Thanks for the reply. I don't understand why that information should have
to be set in sip.conf. It is already known. If I set the following on an
extension:
exten => 100,1,SetCallerID(${CALLERIDNUM})
I can see that the information is correct:
-- Executing SetCallerID("SIP/228-a94f", "228") in new stack
-- Executing Dial("SIP/228-a94f", "SIP/100|20") in new stack
-- Called 100
But, the "number" when extension 100 rings, it gets "100" as the number
portion of the caller ID information.
John
William Suffill wrote:
u probably should upgrade to 0.7.2 but as far as the caller id that
would be from your sip.conf being set improperly add to your sip.conf
callerid="Caller Name" <#> for each sip entry and that should clear it
up.
On Sat, 2004-02-07 at 00:23, John Fraizer wrote:
I'm running Asterisk 0.5.0 and using Cisco 7960 phones in a "sip only"
configuration currently. Everything is working except that caller ID is hosed.
Say for example extension 100 calls extension 200. 200 sees "100" as the
name but "200" as the number. IE, it gets its own number as the supposed
CLID of the calling party.
This is flat out wrong. Am I doing something wrong or is Asterisk just
terribly broken with respect to sending caller ID information properly?
Is this something that only effects Cisco phones?
Thanks,
John
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Re: [Asterisk-Users] Caller-ID is being sent wrong. How to fix it?
u probably should upgrade to 0.7.2 but as far as the caller id that would be from your sip.conf being set improperly add to your sip.conf callerid="Caller Name" <#> for each sip entry and that should clear it up. On Sat, 2004-02-07 at 00:23, John Fraizer wrote: > I'm running Asterisk 0.5.0 and using Cisco 7960 phones in a "sip only" > configuration currently. Everything is working except that caller ID is hosed. > > Say for example extension 100 calls extension 200. 200 sees "100" as the > name but "200" as the number. IE, it gets its own number as the supposed > CLID of the calling party. > > This is flat out wrong. Am I doing something wrong or is Asterisk just > terribly broken with respect to sending caller ID information properly? > > Is this something that only effects Cisco phones? > > Thanks, > > John > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
