A

Anonymous

Question

Use FMG with sip carrier

Forum|Forum|15 years ago
April 19, 2011
2 replies
1577 views

Good day!!! I've installed FMS 4 and FMG, also i configured FMS and FMG to make call to mobile and stationary phone. I use sipnet.ru as SIP-carrier, entered all data to sip.xml in sipGateway

Login: my_login

Pass: my_pass

remoteHost: sipnet.ru

Codec: G711u

after that, i've launched both servers and logged in, this is good. When i try to call to mobile, I see how the call button is changing on the another state, but i dont get calling on my phone.

This carrier gives the special number to find out the latest news and etc. When i try to enter those numbers it works, i hear news and i see how many bytes i received and sent, But still can call to mobile

I downloaded special softphone and entered all data to it, it works, but flash doesn't want to call

Where's my mistake?

p.s.: I used flashphone which i've found in FMG folder

Media gateway

This topic has been closed for replies.

A

Anonymous

When i try to call via flashphoe i see in below textfield that "call" and "reset", when i clicked the 'call' button, the last one hided and i saw another panel with mic, state of calling, etc, when the percenteges went up to 6% i got reset and the panel was changed on the button 'call'. Unfortunately, i didn't see the log file

upd:

There are logs from textfield

Subscribing to CallLeg Stream: fmg/telephony/4

PushNewStream: 1000

Codec:NellyMoser

Audio Rate:8

NetStream.Publish.Start

NetStream.Play.Reset

NetStream.Play.Start

NetStream.Play.PublishNotify

received leg status:leg.state.init

received leg status:leg.state.ring

received leg status:leg.state.sendrecv

Received leg indication:message.indication.progress

NetStream.Play.UnpublishNotify

Received leg indication:message.indication.progress

received leg status:leg.state.hangup

NetStream.Play.Stop

I can't understand why after calling number i got hangup

A

Anonymous

Hi,

From the last indication message that your flash client received; It message.indication.progress means that FMG has stared (or tried) taking action by placing a SIP call to the SIPGateway. Since the call was soon hangup; the a likelihood, the hangup was received by the received from the remoteSIP service provider.

Before moving ahead; following steps would help find out the reason/code sent which remote SIP Server woul have sent before rejecting the outgoing call.

1) locate conf/sip.xml in FMG installation folder; Open the file And locate <debugSIP> which is set to false. Setting this flag to true would include SIP messages in the log/SIPLog_<date/time>.log file. Change the value to true i.e.

2) Restart FMG.

3) Try placing a call again using the flash phone so that FMG generates some logs relevant to the problem you are facing.

4) Locate & open the latest log/SIPLog_<date>.log file. In this file look for last few messages with pattern “SIP/2.0”. Each encounter of pattern would point to a SIP packet sent or received from the remote SIP server; each matching text line would also contain text like INVITE,TRYING, CANCEL etc. A message received from right after “INVITE “ & “TRYING” would explain the cause as well as error code number (e.g. 404). You can match these codes with a reference list here: http://en.wikipedia.org/wiki/List_of_SIP_response_codes

If in doubt while interpreting the code; send across all lines matching with pattern SIP/2.0.

I hope it helps.

-Pankaj

A

Anonymous

Pankaj Y, thanks for your reply. I've done all you adviced, but SIPlog file has a lot of line and it's hard to find something useful there. I set up wireshark and it showed that i got 2 errors (401 and 488), i checked it out and 401 means that my UC isn't authorized, but i can't understand how to solve it??? i filled sipGateway profile in sip.xml with login, password, siphost and other, wehere should i authorize else?

And what error 488 means?

No.     Time        Source                Destination           Protocol Info
     14 1.149627    10.0.2.15             212.53.40.40          SIP/SDP  Request: INVITE sip:79262116048@212.53.40.40:5060, with session description

Frame 14: 690 bytes on wire (5520 bits), 690 bytes captured (5520 bits)
Ethernet II, Src: CadmusCo_42:82:cc (08:00:27:42:82:cc), Dst: RealtekU_12:35:02 (52:54:00:12:35:02)
Internet Protocol, Src: 10.0.2.15 (10.0.2.15), Dst: 212.53.40.40 (212.53.40.40)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol

No.     Time        Source                Destination           Protocol Info
     15 1.157864    212.53.40.40          10.0.2.15             SIP      Status: 100 Trying

Frame 15: 402 bytes on wire (3216 bits), 402 bytes captured (3216 bits)
Ethernet II, Src: RealtekU_12:35:00 (52:54:00:12:35:00), Dst: CadmusCo_42:82:cc (08:00:27:42:82:cc)
Internet Protocol, Src: 212.53.40.40 (212.53.40.40), Dst: 10.0.2.15 (10.0.2.15)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol

No.     Time        Source                Destination           Protocol Info
     16 1.157911    212.53.40.40          10.0.2.15             SIP      Status: 401 Authentication required

Frame 16: 553 bytes on wire (4424 bits), 553 bytes captured (4424 bits)
Ethernet II, Src: RealtekU_12:35:00 (52:54:00:12:35:00), Dst: CadmusCo_42:82:cc (08:00:27:42:82:cc)
Internet Protocol, Src: 212.53.40.40 (212.53.40.40), Dst: 10.0.2.15 (10.0.2.15)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol

No.     Time        Source                Destination           Protocol Info
     17 1.164715    10.0.2.15             212.53.40.40          SIP      Request: ACK sip:79262116048@212.53.40.40:5060

Frame 17: 446 bytes on wire (3568 bits), 446 bytes captured (3568 bits)
Ethernet II, Src: CadmusCo_42:82:cc (08:00:27:42:82:cc), Dst: RealtekU_12:35:02 (52:54:00:12:35:02)
Internet Protocol, Src: 10.0.2.15 (10.0.2.15), Dst: 212.53.40.40 (212.53.40.40)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol

No.     Time        Source                Destination           Protocol Info
     18 1.168727    10.0.2.15             212.53.40.40          SIP/SDP  Request: INVITE sip:79262116048@212.53.40.40:5060, with session description

Frame 18: 940 bytes on wire (7520 bits), 940 bytes captured (7520 bits)
Ethernet II, Src: CadmusCo_42:82:cc (08:00:27:42:82:cc), Dst: RealtekU_12:35:02 (52:54:00:12:35:02)
Internet Protocol, Src: 10.0.2.15 (10.0.2.15), Dst: 212.53.40.40 (212.53.40.40)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol

No.     Time        Source                Destination           Protocol Info
     19 1.176513    212.53.40.40          10.0.2.15             SIP      Status: 100 Trying

Frame 19: 402 bytes on wire (3216 bits), 402 bytes captured (3216 bits)
Ethernet II, Src: RealtekU_12:35:00 (52:54:00:12:35:00), Dst: CadmusCo_42:82:cc (08:00:27:42:82:cc)
Internet Protocol, Src: 212.53.40.40 (212.53.40.40), Dst: 10.0.2.15 (10.0.2.15)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol

No.     Time        Source                Destination           Protocol Info
     20 1.228778    212.53.40.40          10.0.2.15             SIP      Status: 488 Not acceptable here

Frame 20: 432 bytes on wire (3456 bits), 432 bytes captured (3456 bits)
Ethernet II, Src: RealtekU_12:35:00 (52:54:00:12:35:00), Dst: CadmusCo_42:82:cc (08:00:27:42:82:cc)
Internet Protocol, Src: 212.53.40.40 (212.53.40.40), Dst: 10.0.2.15 (10.0.2.15)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol

No.     Time        Source                Destination           Protocol Info
     21 1.232932    10.0.2.15             212.53.40.40          SIP      Request: ACK sip:79262116048@212.53.40.40:5060

Frame 21: 455 bytes on wire (3640 bits), 455 bytes captured (3640 bits)
Ethernet II, Src: CadmusCo_42:82:cc (08:00:27:42:82:cc), Dst: RealtekU_12:35:02 (52:54:00:12:35:02)
Internet Protocol, Src: 10.0.2.15 (10.0.2.15), Dst: 212.53.40.40 (212.53.40.40)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol

Also i see, that it uses UDP, i don't remember, but probably my sip carrier prefers TCP, but i'm not sure

A

Anonymous

Since you are able to place a call on to latest news number of the SIP provider; I am more inclined to assume that your setup is almost ready.

Normally, the call button should disappear while placing a call.

Flash Phone displays last messages & indication received while making a call. These messages are also printed in text window below call button in the Flash Phone. Most of these messages are self descriptive; more information about these messages is available in file: /doc/FMGLegServiceAPI.pdf in the FMG installation folder.

While calling the mobile number, if flash phone never reaches to a state of message.indication.remoteRinging/message.indication.remoteAnswer; then the best way is to dig into FMG/log/core00.log file and see if FMG is really trying to place a call to the mobile number entered by you; further logs would show information about the progress of call and the stage till which call progressed. Some of this information should help you pinpoint the unexpected.

-Pankaj

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