Help - May Asterisk replace a Cisco ata 186?

Asterisk Asterisk Support
1

I am new with Asterisk and I don’t know how to configure Asterisk, unders H323 or Sip, to get in / out calls to the internal extensions without use the Cisco ata 186.

May someone help me with channel configuration and its corresponding in Sip and Extesnions Files.

2

The caseis this:

  1. We have a Asterisk PBX working with one FXO / FXS
  2. All the desk phone are IP Phones or SoftPhones (10)
  3. All incoming and outgoing calls from and to PSTN work well.
  4. Now a VOIP company is offering two more lines for the customer care department.
  5. Informations given us is the Ip Address (for the Provider) and the DID numbers. also this company handle SIP and H323 Protocols

Ok. Right now this company offers the ciscio 18x, for gateway, but it isn’t an “intelligent” solution.

How could be configured Asterisk to accept and send call from this VOIP Provaider?

I read WIKI and seems like a IAX or SIP peer configuration, but we receibve this from the console “Got SIP response 405 “Method Not Allowed” back from …” and later this other " NOTICE[2078]: chan_sip.c:4027 sip_reg_timeout: – Registration for '305…"

Someone knows how to figure out this?

Thanks. (Sorry the enghish I hope this message be readable :frowning:

3

After reading, each Sip and AIX providers must be defined in sip.conf, register => …, also extension.conf must include the sip and iax extesnions,… but income calls are rejected.

When you dial to the DID, you get a busy tone on the phone (caller), Asterisk reports this message: chan_sip.c:2773 process_sdp: No compatible codecs!.. and ethereal presents this analisis:

Session Initiation Protocol
Request-Line: INVITE sip:5034219262@21.24.141.42:5060 SIP/2.0
Method: INVITE
Resent Packet: False
.
.
From: sip:4595609815@21.24.143.26;tag=A6D3C6E4-2263
SIP from address: sip:4595609815@21.24.143.26
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
…
Content-Type: application/sdp
Content-Length: 235
Message body
…
Owner/Creator, Session Id (o): CiscoSystemsSIP-GW- UserAgent 8471 3560 IN IP4 21.24.143.26
Owner Username: CiscoSystemsSIP-GW-UserAgent
… Session Name (s): SIP Call
Media Description, name and address (m): audio 18986 RTP/AVP 4 19
Media Proto: RTP/AVP
Media Format: ITU-T G.723
Media Format: Comfort noise (old)
…
Media Attribute Value: 4 G723/8000

        Media Attribute (a): rtpmap:19 CN/8000
            Media Attribute Fieldname: rtpmap
            Media Attribute Value: 19 CN/8000

And Asterisk responds with this:

Session Initiation Protocol
Status-Line: SIP/2.0 488 Not Acceptable Here
Status-Code: 488
Resent Packet: False
Message Header
Via: SIP/2.0/UDP 21.24.143.26:5060
From: sip:4595609815@21.24.143.26;tag=A6D3C6E4-2263
SIP from address: sip:4595609815@21.24.143.26
SIP tag: A6D3C6E4-2263
To: sip:3054216292@21.24.141.42;tag=as49c97a33
SIP to address: sip:3054216292@21.24.141.42
SIP tag: as49c97a33
Call-ID: 79DFCDFE-482411D5-B926B9D8-C3D3E8A0@21.24.143.26
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:

Finally. can any Sip provider assign a DID, via SIP, and Asteririk receive and route income calls (out going calls also) without any previous REGISTRY ??? How to configure this function? Is this a GATEWAY schema???..

Thank in advance for your help.