[Freeswitch-users] Why doesn't this call get answered?

Fri Apr 16 17:07:46 UTC 2021

I hate to be needy, but does anyone see any reason why I get no answer to
this invite? (I haven't yet generated an invite that works. I guess that
will be my next task.)

INVITE sip:1<outside_number>@us-west-or.sip.flowroute.com SIP/2.0
Via: SIP/2.0/UDP <ext_IP>:5089;rport;branch=z9hG4bKcBHrQKXgHNctm
Max-Forwards: 69
From: "East West Fax" <sip:1<my_CID>@<ext_IP>>;tag=c99SXeSa6mSSQ
To: <sip:1<outside_number>@us-west-or.sip.flowroute.com>
Call-ID: dec531ef-1817-123a-9b91-10e7c6b0315b
CSeq: 34666784 INVITE
Contact: <sip:gw+flowroute@<ext_IP>:5089;transport=udp;gw=flowroute>
User-Agent: FreeSWITCH-mod_sofia/1.10.6-release-18-1ff9d0a60e~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 477
P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=84-D8-1B-E9-50-5F
P-Emergency-Info: IEEE-EUI-48;eui-48-addr=C0-74-AD-30-CD-39
X-FS-Support: update_display,send_info
Alert-Info: <internal>
Remote-Party-ID: "East West Fax"
<sip:1<my_CID>@<ext_IP>>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1618413436 1618413437 IN IP4 <ext_IP>
s=FreeSWITCH
c=IN IP4 <ext_IP>
t=0 0
m=audio 27716 RTP/AVP 0 8 102 9 101 103
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:102 opus/48000/2
a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000;
maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40; stereo=1
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:103 telephone-event/48000
a=fmtp:103 0-16
a=ptime:20

-- 
Steve

On Wed, Apr 14, 2021 at 6:19 PM Steven Schoch <
schoch+freeswitch.org at xwin32.com> wrote:

> The sip trace is attached.
> It seems to show that it sends INVITE messages, but never gets a response.
> However, when it sends an OPTIONS message, it does get a response.
> When calling from a different extension (using a Polycom instead of a
> Grandstream ATA), the INVITE gets answered and the call proceeds.
>
> It seems that there is something "wrong" with this INVITE that makes
> Flowroute ignore it. What could that be, and how do I fix it?
>
> I may have enough data here to ask Flowroute directly, so I'm going to
> give that a try as well.
>
> --
> Steve
>
> On Wed, Apr 14, 2021 at 12:43 PM Brian West <brian at freeswitch.com> wrote:
>
>> That's the FMTP for OPUS.  Chances are that invite breaks the Polycom,
>> what does the SDP look like coming back from that invite? I'll be you, it's
>> broken.
>>
>> /b
>>
>>
>> On Wed, Apr 14, 2021 at 1:51 PM Bote Man <botelist at gmail.com> wrote:
>>
>>> I am absolutely no expert on SDP, but that SDP line that begins
>>>
>>> a=fmtp:102 useinbandfec=1…
>>>
>>> looks to me like it’s trying to set up a video call. I saw this behavior
>>> with the newer Polycom VVX501 before I beat those eager beavers into
>>> submission.
>>>
>>>
>>> It also looks like the Grandstream is offering a lot more codecs which
>>> you might prefer to trim down to only those necessary to get the job done.
>>> Sometimes additional codecs or codecs listed in the “wrong” sequence can
>>> cause mystery problems.
>>>
>>>
>>>
>>> You might have to resort to siptrace logging between FS and your carrier.
>>>
>>> sofia profile external siptrace on ç or whatever profile handles your
>>> provider; or maybe internal to snoop what’s going on between FS and your
>>> Grandstream.
>>>
>>>
>>>
>>> Hope this helps.
>>>
>>>
>>>
>>>
>>>
>>> ---
>>>
>>> John Boteler
>>>
>>> BnC Group U.S.A.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org>
>>> *On Behalf Of *Steven Schoch
>>> *Sent:* Tuesday, 13 April, 2021 20:47
>>> *To:* freeswitch-users <freeswitch-users at lists.freeswitch.org>
>>> *Subject:* [Freeswitch-users] Why doesn't this call get answered?
>>>
>>>
>>>
>>> This office has a bunch of Polycom SoundPoint IP 320 phones, and a
>>> single Grandstream HT801 (connected to a FAX machine for outgoing FAXes).
>>>
>>>
>>>
>>> I can make a call from the phones to outside numbers.
>>>
>>> I can make a call from the HT801 to local phones.
>>>
>>> But I can't call from the HT801 to outside numbers.
>>>
>>>
>>>
>>> The last important thing that happens in the failed call is this:
>>>
>>> 2021-04-13 17:28:15.260844 [DEBUG] sofia.c:7406 Channel
>>> sofia/external/<number> entering state [calling][0]
>>>
>>>
>>>
>>> The difference between the work and not work seems to be this: When I
>>> call from a phone to an outside number, it does this:
>>>
>>>
>>>
>>> Local SDP:
>>> v=0
>>> o=FreeSWITCH 1618335260 1618335261 IN IP4 <external-IP>
>>> s=FreeSWITCH
>>> c=IN IP4 <external-IP>
>>> t=0 0
>>> m=audio 25104 RTP/AVP 0 8 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>> a=sendrecv
>>>
>>>
>>>
>>> When it tries to call from the HT801 to an outside number, it does this:
>>>
>>>
>>>
>>> Local SDP:
>>>
>>> v=0
>>> o=FreeSWITCH 1618327543 1618327544 IN IP4 <external-IP>
>>> s=FreeSWITCH
>>> c=IN IP4 <external-IP>
>>> t=0 0
>>> m=audio 32552 RTP/AVP 0 8 102 9 101 103
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:102 opus/48000/2
>>> a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000;
>>> maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40; stereo=1
>>> a=rtpmap:9 G722/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=rtpmap:103 telephone-event/48000
>>> a=fmtp:103 0-16
>>> a=ptime:20
>>> a=sendrecv
>>>
>>>
>>>
>>> Is that why it doesn't answer? If so, how do I change it?
>>>
>>>
>>>
>>> I should mention that when I tried this at home, it worked, but when I
>>> attempted to install it here at the bookstore, it didn't. The Comcast
>>> router at my home is a little different; they use a Netgear router here;
>>> and I may have upgraded Freeswitch between the time it worked and now.
>>>
>>>
>>>
>>> --
>>>
>>> Steve
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
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>>>
>>> FreeSWITCH-users mailing list
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>>> https://freeswitch.com
>>
>>
>>
>> --
>>
>> Brian West | Co-founder and Developer
>>
>> Need Commercial support? email sales at freeswitch.com
>>
>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
>> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>>
>> Email: brian at freeswitch.com
>>
>> Mobile: 918-424-9378
>>
>> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>>
>> [image: https://www.facebook.com/signalwireinc?src=email]
>> <https://www.facebook.com/freeswitch> [image:
>> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
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