[Asterisk-Dev] Phoneserve SIP provider

[Sergey S. Stasyuk]

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
>  
> ь

>  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