Re: [asterisk-users] ISP trunk session ID [SOLVED]

Sergej Petrovsky Sun, 12 May 2013 05:50:11 -0700

Hello

I solved it and I leave the solution here for those who need it because this modems software is retarded, it does not allow to put the VOIP (VSPA_SIP) client to any other interface so I couldn't do the trick to connect it back to my own asterisk server on LAN and monitor what packets it send out. It only connects through the ONT (optical) link through it's wan adapter, and that cannot be bridged again. So basically I can't even use it as an ATA later for my own pbx...

So I needed to register at the PBX in this format:

instead of:

Best Regards,

Sergej

11.05.2013, 10:30, "Sergej Petrovsky" <sergej5...@yandex.com>:

Hello

Thanks for the response! I already set both Agents to empty (also tried the username), doesn't make any difference. What I could figure out from the modem's SIP client is:

#grep -ri agent /bin/vspa_sip

HW_SIP_FormatUserAgentStr
SIPPA_FunFillUserAgentHead
HW_VSPA_CheckUriAndUserAgentDomain
HW_VSPA_CheckUserAgentDomain
ParseUserAgentMemCp
SipUaDlgUAddUserAgentAndOrgnizationHeaders
SipUaUtilAddUserAgentHeader
SipUserIeIniUserAgent
HW_VSPA_CheckUserAgentDomain Failed.ulRet=0x%x.
UserAgent
UserAgentDomain Check Err!VoiceProfile.{%u}.Line.{%u}.URI=%s,aucUserAgentDomain=%s
[VOIP] %s VoiceProfile[%lu] Line[%lu] UserAgentDomain Changed From %s to %s.
User Agent:%s
SipLmSetSoftConfigPara SIP_SOFT_CONFIG_ADD_USER_AGENT_FOR_ALL_UA_MSGS failed.ulRet=%u
Feature: LOG/TRACE/STATISTICS/BACKUP/IPV4_SUPPORT/OPTIMIZE_APPMSG/MIB_STATISTICS/ETAG/ADD_USER_AGENT_HDR/32_BIT/
User-Agent
User-Agent:
bInsUserAgent: %u
SipUaUtilAddUserAgentHeader
SipUaDlgUAddUserAgentAndOrgnizationHeaders
Add UserAgent and Organization header failed

The problem really is that it register and from that point you have no way knowing what goes wrong on the other side...

Sergej

11.05.2013, 09:50, "Asghar Mohammad" <asghar...@gmail.com>:
you can find in [general] section.
useragent=asterisk ; Allows you to change the user agent string
; The default user agent string also contains the Asterisk
; version. If you don't want to expose this, change the
; useragent string.
sdpsession=asterisk ; Allows you to change the SDP session name string, (s=)
; Like the useragent parameter, the default user agent string
; also contains the Asterisk version.
;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)

On Sat, May 11, 2013 at 5:16 AM, Nick Khamis <sym...@gmail.com> wrote:
Sorry to chime in here, is it possible to change the "Server: Asterisk
", "s=Asterisk", and "o=" within sip.conf? What are the directives
exactly please?

Thanks in Advance,

Nick.

On 5/10/13, Asghar Mohammad <asghar...@gmail.com> wrote:
> hi,
> you can try to change sip user agent and sdp session s , owner in sip
> config same as your phone,s (modem).
> asterisk by default send user agent = asterisk version , s= asterisk , o=
> asterisk.
> some providers are not happy if they see "asterisk" word :)
>
>
>
> On Sat, May 11, 2013 at 12:27 AM, Sergej Petrovsky
> <sergej5...@yandex.com>wrote:
>
>> Hi folks,
>>
>> What I trying to do here is exactly this:
>> http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt599.html
>>
>> My provider given me a Huawei modem which have 2 phone jacks on it, but
>> instead of using it I rather redirect my POTS number to my PBX. I ran
>> into
>> couple of bumps on the road but now it's "half-working". I extracted the
>> SIP user, pass, server info from the modem and even managed to put my PBX
>> into the same VLAN they use, on the exact same IP address like the modem
>> but there is 1 problem:
>> It seems this modem also sends some session ID to the ISP's sip server,
>> something what Asterisk doesn't by default. So if I do this:
>>
>> 1, Let the modem register at the sip service (the phone number can be
>> called and ringing out)
>> 2, Disconnect the modem
>> 3, Let the PBX connect to the SIP server
>> 4, PBX accepts the calls
>> 5, About 5-10 minutes later it stops doing it, when I call the number it
>> shows busy (beep, beep, beep), no matter if I restart Asterisk or not it
>> won't work anymore just if I do the same trick again
>>
>> I'm sure the remote SIP server breaks the voip channel or something, it
>> does NOT drop me out tho, my PBX can register any time without problem
>> but
>> no packets will ever come forward me anymore. It's kind of hard to solve
>> this from 1 side.
>>
>> There must be some solution for this.
>>
>> Please help!
>>
>> Thank You,
>> Sergej
>>
>>
>>
>> --
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>

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Re: [asterisk-users] ISP trunk session ID [SOLVED]

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