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

Wed Apr 14 21:10:54 UTC 2021

Calls to the Polycom are not broken - that works. To summarize:

Analog telephone -> HT801 -> Freeswitch -> Polycom SoundPoint works.
Polycom SoundPaint -> Freeswitch -> Flowroute -> mobile phone (or any PSTN
number) works.
Analog telephone -> HT801 -> Freeswitch -> Flowroute -> any PSTN number
fails.

As John pointed out, it could be the Grandstream codecs are causing
Flowroute to ignore the call. I will have to try siptrace.

-- 
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
>> _________________________________________________________________________
>>
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>
>
>
> --
>
> Brian West | Co-founder and Developer
>
> Need Commercial support? email sales at freeswitch.com
>
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> _________________________________________________________________________
>
> 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
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