[Asterisk-Dev] Phoneserve SIP provider

[Martin Pycko]

try regiser => user:pass at ip

not

register => user at ip

Martin

On 15 Jul 2003, Sergey S. Stasyuk wrote:

> Hi all!
>
> I am using asterisk in local net (only SIP), all looks good from ATA-186
> and eStara SoftPhone.
> Before I even connects with it to PhoneServe provider, but now I can't
> register with it. Asterisk tries to register, but:
>
>
> > voip*CLI> sip debug
> > SIP Debugging Enabled
> > 11 headers, 0 linesregi
> > Reliably Transmitting:
> > REGISTER sip:sip186.phoneserve.com SIP/2.0
> > Via: SIP/2.0/UDP 212.86.112.165:5060;branch=z9hG4bK0afbed32
> > From: <sip:NNNNNNNNNNNNNNNNNN at sip186.phoneserve.com>;tag=as11e39a0c
> > To: <sip:NNNNNNNNNNNNNNNNNN at sip186.phoneserve.com>
> > Call-ID: 0e2a028f46a9e01a68139aa70a43eabb at 212.86.112.165
> > CSeq: 103 REGISTER
> > User-Agent: Asterisk PBX
> > Expires: 120
> > Contact: <sip:NNNNNNNNNNNNNNNNNN at 212.86.112.165>
> > Event: registration
> > Content-length: 0
> >
> >  (no NAT) to 62.189.6.93:5060
> > Retransmitting #1 (no NAT):
> > REGISTER sip:sip186.phoneserve.com SIP/2.0
> > Via: SIP/2.0/UDP 212.86.112.165:5060;branch=z9hG4bK0afbed32
> > From: <sip:NNNNNNNNNNNNNNNNNN at sip186.phoneserve.com>;tag=as11e39a0c
> > To: <sip:NNNNNNNNNNNNNNNNNN at sip186.phoneserve.com>
> > Call-ID: 0e2a028f46a9e01a68139aa70a43eabb at 212.86.112.165
> > CSeq: 103 REGISTER
> > User-Agent: Asterisk PBX
> > Expires: 120
> > Contact: <sip:NNNNNNNNNNNNNNNNNN at 212.86.112.165>
> > Event: registration
> > Content-length: 0
> >
> > Ø

> >  to 62.189.6.93:5060
> ..... So on to Retransmitting #5
>
> My sip.conf includes:
>
> > register => NNNNNNNNNNNNNNNNNN at sip186.phoneserve.com
>
> > [phoneserveprovider]
> > type=friend
> > username=NNNNNNNNNNNNNNNNNN
> > host=62.189.6.93
> > fromdomain=sip186.phoneserve.com
> > canreinvite=no
>
> extensions.conf:
>
> > [sip]
> > exten => 7777,1,DateTime
> > exten => 7777,2,Hangup
> > exten => 7777,t,Hangup
> >
> > exten => _XXXX,1,Dial(SIP/${EXTEN})
> > exten => _XXXX,t,Hangup
> >
> > ; Transfer call to owr Service Provider
> > exten => _XXXXXXXXXXXXXXXXXX,1,Dial(SIP/${EXTEN}@SIP186.PHONESERVE.COM)
> > exten => _XXXXXXXXXXX,1,Dial(SIP/${EXTEN}@SIP186.PHONESERVE.COM)
>
> By the way, phoneserve.com ATA-186 users have to use two different accounts.
> How to configure extensions to use second account only if first is busy.
>
> Phone1 |\                                  /| PhoneServe account 1
>          \|                              |/
>           | ATA-186 |-----| Asterisk Box |
>          /|                      |       |\
> Phone2 |/                        |         \| PhoneServe account 2
>                                  |
>                           Non-ATA users
>
> Best reagrds,
> Sergey Stasyuk
>
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