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

Fri Apr 16 18:48:29 UTC 2021

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
>>
>>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20210416/514108e0/attachment-0001.html>