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

Steven Schoch schoch+freeswitch.org at xwin32.com
Fri Apr 16 21:16:14 UTC 2021


I configured the ATA to offer only 2 codecs, which made it get further,
until Flowroute asked for authentication, which then made the next packet
too big.
So I then found another setting in the ATA to stop using
the P-Access-Network-Info and P-Emergency-Info headers. This made the
packet small enough to complete the call.

Success!

I will ask Flowroute if big packets are dropped on their end, or if it's a
problem with the Netgear UDP NAT routing.

-- 
Steve

On Fri, Apr 16, 2021 at 1:17 PM Guillermo Ruiz Camauer <grcamauer at gmail.com>
wrote:

> To test if that is true, offer less Codecs.  That will shrink the 9 bytes
> you need.
>
> Guillermo
>
> On Fri, Apr 16, 2021 at 4:33 PM Steven Schoch <
> schoch+freeswitch.org at xwin32.com> wrote:
>
>> I have a theory.
>>
>> The INVITE that originates from the HT801 is bigger, and results in a UDP
>> packet of 1509 bytes vs 1157 bytes for the one that works.
>>
>> The MTU for Ethernet is 1500, which means the larger UDP packet will get
>> fragmented. Maybe the Netgear router is not handling fragmented UDP packets
>> properly, or maybe the Linux system is sending a jumbo frame and the
>> Netgear router is dropping it. I will investigate...
>>
>> --
>> Steve
>>
>> On Fri, Apr 16, 2021 at 11:48 AM Steven Schoch <
>> schoch+freeswitch.org at xwin32.com> wrote:
>>
>>> Here's the one that works. One difference I notice is that the one that
>>> works has a 10-digit Caller-ID, where the one that doesn't work has an
>>> 11-digit Caller-ID (starting with 1). There are other differences as well:
>>>
>>> send 1157 bytes to udp/[34.210.91.114]:5060 at 11:38:10.114622:
>>>
>>> ------------------------------------------------------------------------
>>>
>>> INVITE sip:1<outside_number>@us-west-or.sip.flowroute.com SIP/2.0
>>>
>>> Via: SIP/2.0/UDP <ext_IP>:5089;rport;branch=z9hG4bKXDQ05vU1NB3yD
>>>
>>> Max-Forwards: 69
>>>
>>> From: "East West Bookshop" <sip:<my_CID>@<ext_IP>>;tag=gymQS2XU0H79e
>>>
>>> To: <sip:1<outside_number>@us-west-or.sip.flowroute.com>
>>>
>>> Call-ID: bc02d323-1985-123a-9b91-10e7c6b0315b
>>>
>>> CSeq: 34745353 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: 244
>>>
>>> X-FS-Support: update_display,send_info
>>>
>>> Alert-Info: <internal>
>>>
>>> Remote-Party-ID: "East West Bookshop"
>>> <sip:<my_CID>@<ext_IP>>;party=calling;screen=yes;privacy=off
>>>
>>>
>>> v=0
>>>
>>> o=FreeSWITCH 1618573488 1618573489 IN IP4 <ext_IP>
>>>
>>> s=FreeSWITCH
>>>
>>> c=IN IP4 <ext_IP>
>>>
>>> t=0 0
>>>
>>> m=audio 24802 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
>>>
>>>
>>>
>>>
>>> recv 336 bytes from udp/[34.210.91.114]:5060 at 11:38:10.172288:
>>>
>>> ------------------------------------------------------------------------
>>>
>>> SIP/2.0 100 Trying
>>>
>>> Via: SIP/2.0/UDP
>>> <ext_IP>:5089;rport=5089;branch=z9hG4bKXDQ05vU1NB3yD;received=<ext_IP>
>>>
>>> From: "East West Bookshop" <sip:<my_CID>@<ext_IP>>;tag=gymQS2XU0H79e
>>>
>>> To: <sip:1<outside_number>@us-west-or.sip.flowroute.com>
>>>
>>> Call-ID: bc02d323-1985-123a-9b91-10e7c6b0315b
>>>
>>> CSeq: 34745353 INVITE
>>>
>>> Content-Length: 0
>>>
>>> --
>>> Steve
>>>
>>> On Fri, Apr 16, 2021 at 10:07 AM Steven Schoch <
>>> schoch+freeswitch.org at xwin32.com> wrote:
>>>
>>>> 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
>>>>>>> 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
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>>
>>>>>> 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
>>>>>
>>>>>
>> _________________________________________________________________________
>>
>> 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
>
>
>
> --
> Guillermo Ruiz Camauer
> _________________________________________________________________________
>
> 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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