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

Fri Apr 16 20:27:35 UTC 2021

You might be right. I have seen it happening with a few carriers. Large
packets get dropped & one sees no response. You can try with just a single
codec (or get rid of some large headers) if this theory is true.

On Fri, Apr 16, 2021 at 3:33 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:
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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/>
>>>>>
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>>>>> <https://www.facebook.com/freeswitch> [image:
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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
>>>>> 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
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