Re: [Asterisk-Users] Polycom 300 setup and AMP
You can edit the polycom 300's via http/your web browser. Or you can set them up to grab their information via ftp. If you are controlling multiple phones I recommend the ftp route. The resource you want to use is: http://www.voip-info.org/wiki-Polycom+Phones On Thu, 2005-05-05 at 06:23, Chris O'Quinn wrote: > Does anyone know where I may be able to find a cheat sheet for setting > up a polycom 300 and AMP - We have set up some software phones, Sipuras > and Budge Tones, but we havent been successful yet with the polycoms. We > do have the polycom instructional pdf , but we need to filter out what > is needed. > > Thank you > C > > __ > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voip-Info
Is anyone else having issues pulling up voip-info.org? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 2100 and Asterisk - Faxing
It would be wonderful if someone would please point me to any good resources on configuring my sipura 2100 and asterisk for faxing. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: ***SOLVED*** [Asterisk-Users] Broadvoice latest changesandstillnot working- An Additional Server****Solved*****!
I had the same problem. But I had a Sip 400 Bad Result ... Failed to authenticate on INVITE ... I am running asterisk 1.03 I do not have my program pointing to any proxies...It is pointing to Sip.broadvoice.com, I do not have any proxies set up in my /etc/hosts file where PP = Phone Number X = secret Try cutting and pasting mine in, see if it works... My Sip.conf is as follows: [general] context=sip ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls dtmfmode=inband disallow=all allow=ulaw allow=gsm register => PP:[EMAIL PROTECTED] [PP] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=PP secret=XX username=PP insecure=very context=sip authname=PP dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no On Thursday 10 March 2005 02:08 pm, Joe wrote: > Zanzamar, > > I agree that it should work. I can call out and have the land phone > ring, but as soon as it is answered, another invite goes out and that > is when I get the 401 not authorized. I don't want to go down this > route, but could this be a Codec issue? > > Here is my sip config > > > > [sip.broadvoice.com] > type=peer > host=sip.broadvoice.com > fromdomain=sip.broadvoice.com > fromuser= BB > username= BB > authuser= BB > secret= secret > context=sip > nat=no > insecure=very > dtmfmode=inband > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: ***SOLVED*** [Asterisk-Users] Broadvoice latest changes and stillnot working- An Additional Server ****Solved*****!
I had this same problem. It was still not authenticating on Invite when you look at it in non sip debug mode, just asterisk -r I found that using just the generic sip.broadvoice.com and making sure I had the right settings in my Sip.conf worked for me On Thursday 10 March 2005 10:37 am, Joe wrote: > Mark and all, > > I rebuilt as well, but I do not have the same results as you. 400 bad > response and 401 not authorized. I dial out just fine, but as soon as > the phone answerers on the other end, I start getting 401 not > authorized and 400 bad response errors and no audio is sent. I can hear > the phone picked up and audio for 1/4 of a second, and then it goes > silent. I hear a clicking sound when I send outbound calls and pick up > the receiving party phone. I'm using the default > proxy.dca.broadvoice.com server. > > Joe > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber > Sent: Wednesday, March 09, 2005 8:34 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: ***SOLVED*** [Asterisk-Users] Broadvoice latest changes and > stillnot working- An Additional Server Solved*! > > I concur. I rebuilt today and now I seem to be able to dial out. > > MARK. > > Chris Nibeck wrote: > > thank you everyone! > > > > It does not seen that it was configuration problems at all. > > > > It appears it was the CVS that I was using from yesterday. > > > > I decided to start over, downloaded the latest CVS, recompiled, and > > voila! * started working > > > > Indeed even a Cisco ATA that was never working before started working! > > > > > > Thanks to everyone! > > > > Chris > > > > On Mar 9, 2005, at 10:28 AM, Chris Nibeck wrote: > >> there are two of us with the same problem so I will answer for me. > >> Yes I tried the below instructions. > >> > >> The current thinking by multiple people is * never tries > >> authenticating so removing the FQDN will force * to go to the related > >> > >> section named by either a phone number or a non Fully Qualified > >> Domain Name. > >> > >> But I still don't have it working so who knows. > >> > >> Anyone that wishes to call me via BV my number is 8475100139 and it > >> is up. > >> > >> Chris > >> > >> On Mar 9, 2005, at 9:23 AM, Mike Matthews wrote: > >>> Have you tried this: > >>> > >>> http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup > >>> > >>> Zanzamar Majere wrote: > >>>> Thank you for the response. I still have the errors mentioned > >>>> below, sip response and Failed to authenticate on INVITE > >>>> > >>>> [PP] > >>>> type=peer > >>>> username=PP > >>>> fromuser=PP > >>>> authuser=PP > >>>> fromdomain=sip.broadvoice.com > >>>> secret=XX > >>>> host=sip.broadvoice.com > >>>> dtmfmode=inband > >>>> insecure=very > >>>> context=sip > >>>> qualify=yes > >>>> disallow=all > >>>> allow=ulaw > >>>> allow=gsm > >>>> ;Disable canreinvite if you are behind a NAT > >>>> ;canreinvite=no > >>>> nat=no > >>>> > >>>> Does anyone else have any other suggestions? > >>> > >>> ___ > >>> Asterisk-Users mailing list > >>> Asterisk-Users@lists.digium.com > >>> 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 > >> Asterisk-Users@lists.digium.com > >> 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 > > Asterisk-Users@lists.digium.com > > 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 > Asterisk-Users@lists.digium.com > 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 > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Fwd: Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server ****SOLVED****
This configuration solved my problem. I could have sworn I tried this before. I guess not. I did not need to apply the patch. Also, I am using a regular Registration setup in my sip.conf not broadvoice's funky one... The only thing I can surmise is that order of the variables matters. This is what worked for me: [PP] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=PP secret=XX username=PP insecure=very context=sip authname=PP dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no Thank you On Wednesday 09 March 2005 08:23 am, Mike Matthews wrote: > Have you tried this: > > http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup > > Zanzamar Majere wrote: > >Thank you for the response. I still have the errors mentioned below, sip > >response and Failed to authenticate on INVITE > > > >[PP] > >type=peer > >username=PP > >fromuser=PP > >authuser=PP > >fromdomain=sip.broadvoice.com > >secret=XX > >host=sip.broadvoice.com > >dtmfmode=inband > >insecure=very > >context=sip > >qualify=yes > >disallow=all > >allow=ulaw > >allow=gsm > >;Disable canreinvite if you are behind a NAT > >;canreinvite=no > >nat=no > > > >Does anyone else have any other suggestions? > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com 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] Broadvoice latest changes and still not working- An Additional Server
Thank you for the response. I still have the errors mentioned below, sip
response and Failed to authenticate on INVITE
[PP]
type=peer
username=PP
fromuser=PP
authuser=PP
fromdomain=sip.broadvoice.com
secret=XX
host=sip.broadvoice.com
dtmfmode=inband
insecure=very
context=sip
qualify=yes
disallow=all
allow=ulaw
allow=gsm
;Disable canreinvite if you are behind a NAT
;canreinvite=no
nat=no
Does anyone else have any other suggestions?
On Wednesday 09 March 2005 06:56 am, MF Hulber wrote:
> Try changing the extension from Broadvoice1 to the actual phone number
> (and don't send your secret in a public email or maybe that's Chris'):
>
> [*8475100139*]
> type=peer
> ;user=phone
> host=sip.broadvoice.com
> fromdomain=sip.broadvoice.com
> fromuser=8475100139
> secret=XXXXXXX
> username=8475100139
>
> Zanzamar Majere wrote:
> >I have made all the changes to sip.conf for my broadvoice peer
> >friend(and I have tried it as peer) and I am still seeing this response
> >(on call out). Any suggestions? I don't think it is a problem with the
> >phones themselves authenticating, as Asterisk takes care of all the
> >authentication from my understanding.
> >
> >Free world does work for calling out however. So I know at least that
> >works.
> >
> >
> >
> >-- Got SIP response 400 "Bad request" back from 147.135.0.128
> >Mar 9 01:11:09 NOTICE[12284]: chan_sip.c:6798 handle_response: Failed
> >to authenticate on INVITE to '"PP"
> >;tag=as5b80cade'
> >
> >On Wed, 2005-03-09 at 00:48, Chris Nibeck wrote:
> >>First off... please cancel previous amplification request. I have
> >>implemented your ideas with the same errored result.
> >>
> >>I am not sure that we are not making it thru authentication. From my
> >>digging and comparing packet dumps comparing the soft phone to asterisk
> >>they have identical transactions through the ACK reply (the last one
> >>on the debug below). The softphone seems to be authenticated after the
> >>ACK. I am a newbie to debugging this stuff. I just want to get it
> >>working.
> >>
> >>Thanks everyone in advance for your help. I am certainly very very
> >>happy to try anything.
> >>
> >>Based on Luki's suggestions I...
> >>
> >>Changed sip.conf...
> >>
> >>[broadvoice1]
> >>type=peer
> >>;user=phone
> >>host=sip.broadvoice.com
> >>fromdomain=sip.broadvoice.com
> >>fromuser=8475100139
> >>secret=DELETED
> >>username=8475100139
> >>insecure=very
> >>context=default
> >>authname=8475100139
> >>dtmfmode=inband
> >>dtmf=inband
> >>;Disable canreinvite if you are behind a NAT
> >>canreinvite=no
> >>nat=no
> >>
> >>Changed extensions.conf...
> >>
> >>exten => _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice
> >>for 30 seconds
> >>exten => _8X.,2, congestion() ; No answer, nothing
> >>exten => _8X., 102, busy() ;
> >>
> >>End result...
> >>
> >>Mar 9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed
> >>to authenticate on INVITE to '"6050"
> >>;tag=as545ccba3'
> >>
> >>
> >>SIP debug...
> >>
> >> -- Executing Dial("SIP/6050-132b",
> >>"SIP/[EMAIL PROTECTED]|30") in new stack
> >>We're at xxx.xxx.xxx.xxx port 18212
> >>Answering with capability 2
> >>Answering with capability 4
> >>Answering with capability 8
> >>12 headers, 10 lines
> >>Reliably Transmitting:
> >>INVITE sip:[EMAIL PROTECTED] SIP/2.0
> >>Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
> >>From: "6050" ;tag=as545ccba3
> >>To:
> >>Contact:
> >>Call-ID: [EMAIL PROTECTED]
> >>CSeq: 102 INVITE
> >>User-Agent: Asterisk PBX
> >>Date: Wed, 09 Mar 2005 07:30:41 GMT
> >>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> >>Content-Type: application/sdp
> >>Content-Length: 205
> >>
> >>v=0
> >>o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx
> >>s=session
> >>c=IN IP4 xxx.xxx.xxx.xxx
> >>t=0 0
> >>m=audio 18212 RTP/AVP 3 0 8
> >>a=rtpmap:3 GSM/8000
> >>a=rtpmap:0 PCMU/8000
> >>a=rtpmap:8 PCMA/8000
> >>a=silenceSupp:off - - - -
> >> (no NAT) to 147.135.8.128:506
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
I have made all the changes to sip.conf for my broadvoice peer
friend(and I have tried it as peer) and I am still seeing this response
(on call out). Any suggestions? I don't think it is a problem with the
phones themselves authenticating, as Asterisk takes care of all the
authentication from my understanding.
Free world does work for calling out however. So I know at least that
works.
-- Got SIP response 400 "Bad request" back from 147.135.0.128
Mar 9 01:11:09 NOTICE[12284]: chan_sip.c:6798 handle_response: Failed
to authenticate on INVITE to '"PP"
;tag=as5b80cade'
On Wed, 2005-03-09 at 00:48, Chris Nibeck wrote:
> First off... please cancel previous amplification request. I have
> implemented your ideas with the same errored result.
>
> I am not sure that we are not making it thru authentication. From my
> digging and comparing packet dumps comparing the soft phone to asterisk
> they have identical transactions through the ACK reply (the last one
> on the debug below). The softphone seems to be authenticated after the
> ACK. I am a newbie to debugging this stuff. I just want to get it
> working.
>
> Thanks everyone in advance for your help. I am certainly very very
> happy to try anything.
>
> Based on Luki's suggestions I...
>
> Changed sip.conf...
>
> [broadvoice1]
> type=peer
> ;user=phone
> host=sip.broadvoice.com
> fromdomain=sip.broadvoice.com
> fromuser=8475100139
> secret=zjh018g8f8
> username=8475100139
> insecure=very
> context=default
> authname=8475100139
> dtmfmode=inband
> dtmf=inband
> ;Disable canreinvite if you are behind a NAT
> canreinvite=no
> nat=no
>
> Changed extensions.conf...
>
> exten => _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice
> for 30 seconds
> exten => _8X.,2, congestion() ; No answer, nothing
> exten => _8X., 102, busy() ;
>
> End result...
>
> Mar 9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed
> to authenticate on INVITE to '"6050"
> ;tag=as545ccba3'
>
>
> SIP debug...
>
> -- Executing Dial("SIP/6050-132b",
> "SIP/[EMAIL PROTECTED]|30") in new stack
> We're at xxx.xxx.xxx.xxx port 18212
> Answering with capability 2
> Answering with capability 4
> Answering with capability 8
> 12 headers, 10 lines
> Reliably Transmitting:
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
> From: "6050" ;tag=as545ccba3
> To:
> Contact:
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Date: Wed, 09 Mar 2005 07:30:41 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Type: application/sdp
> Content-Length: 205
>
> v=0
> o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx
> s=session
> c=IN IP4 xxx.xxx.xxx.xxx
> t=0 0
> m=audio 18212 RTP/AVP 3 0 8
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> (no NAT) to 147.135.8.128:5060
> -- Called [EMAIL PROTECTED]
> com*CLI>
>
> Sip read:
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
> From: 6050 ;tag=7e2776985d5a0891o0
> To:
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> Max-Forwards: 70
> Proxy-Authorization: Digest
> username="6050",realm="asterisk",nonce="42d82e9b",uri="sip:
> [EMAIL PROTECTED]",algorithm=MD5,response="420e39b35648a10c
> 129dd4fb5f97ec47"
> Contact: 6050
> Expires: 240
> User-Agent: Sipura/SPA3000-2.0.10(GWf)
> Content-Length: 241
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura
> Content-Type: application/sdp
>
> v=0
> o=- 1138990026 1138990026 IN IP4 64.4.192.110
> s=-
> c=IN IP4 64.4.192.110
> t=0 0
> m=audio 16388 RTP/AVP 0 100 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:100 NSE/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:30
> a=sendrecv
>
> 15 headers, 12 lines
> Ignoring this request
> Transmitting (no NAT):
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
> From: 6050 ;tag=7e2776985d5a0891o0
> To: ;tag=as2f065f18
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact:
> Content-Length: 0
>
>
> to 64.4.192.110:5060
> com*CLI>
>
> Sip read:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
> From: "6050" ;tag=as545ccba3
> To:
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
>
>
> 6 headers, 0 lines
> com*CLI>
>
> Sip read:
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
> From: "6050" ;tag=as545ccba3
> To: ;tag=SD38rq699-
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> WWW-Authenticate: DIGEST
> realm="BroadWorks",algorithm=MD5,nonce="1110353299563"
> Content-Length: 0
>
>
> 8 headers, 0 lines
> Transmitting:
> ACK sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
> From: "6050" ;tag=as545ccba3
> To: ;tag=SD38rq699-
> Con
[Asterisk-Users] Sip 400 bad request - broadvoice error
I have searched the list and cannot find a sip 400 solution posted that solves my problem. If anyone has any thoughts or suggestions on the following I would greatly appreciate it. I didn't have this error before Broadvoice made their changes this weekend. Now when I make a call it connects but, I cannot hear anything on the other end... The full message I have is: 8 headers, 0 lines Sending to 147.135.4.128 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 147.135.4.128:5060;branch=z9hG4bK2ib80u200o5079o6q7c1.1sr From: ;tag=SD50bgc99-1538429642-1110330497384 To: "Waiting Room/Caribou Insurance" ;tag=as49ead6ef Call-ID: [EMAIL PROTECTED] CSeq: 40722885 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 147.135.4.128:5060 Destroying call '[EMAIL PROTECTED]' server*CLI> Sip read: SIP/2.0 400 Bad request Via: SIP/2.0/UDP 67.42.244.202:5060;branch=z9hG4bK2d71d419 From: "asterisk" ;tag=as49ead6ef To: ;tag=SD50bgc99-1929245493-1110330503603 Call-ID: [EMAIL PROTECTED] CSeq: 107 INVITE Content-Length: 0 My SIP.conf is: register => @sip.broadvoice.com::@sip.broadvoice.com [zachphone] type=friend [EMAIL PROTECTED] password=.asd.fgh. host=dynamic dtmfmode=inband defaultip=192.168.55.65 mailbox=12 callerid="Zach/Caribou Insurance" <303.557.0057x12> context=sip ;your context in extensions.conf [sip.broadvoice.com] type=friend username=3035570057 fromuser=__ authname=__ fromdomain=sip.broadvoice.com (sip.broadvoice.com is configured in my /etc/hosts to point to proxy.lax.broadvoice.com's ip) secret=-- host=sip.broadvoice.com port=5060 dtmfmode=inband insecure=very context=sip qualify=yes disallow=all allow=ulaw allow=glaw allow=gsm ;Disable canreinvite if you are behind a NAT ;canreinvite=no nat=no ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
