Good morning Christopher,
You know... I'm a technical guy... but sometimes cause and effect eludes
me...
After playing around a while with your suggestions it turns out disabling
"AAC-ELD 16kHz" in the audio settings solves the issue?!?!?
I can understand why a not supported codec can result in a "call failed".
But how is it possible, that an audio setting is OK for local
(inner-country) calls but wrong for international?!?!
Do you have any clue?
Anyhow... thanks heaps for the awesome help with my issues.
Thanks again!
Soko
On Tue, 08 Jan 2019 19:18:14 +0100, Christopher Woods
<[email protected]> wrote:
488 not acceptable here can be due to codec problems. In the SDP in the
initial invite, I don't see any G.711 A-law or U-law (aka PCMA/>PCMU)
listed - only Speex, iLBC, AAC and Opus plus the usual signalling
telephone-event stuff.
Chances are your new telco provider may not wish to transcode to G.711,
so is refusing the set up? Settings -> Audio -> PCMU/PCMA at >the
bottom. I'd expect to see
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
in the SDP from your client along with all the other codecs, if calling
SIP to PSTN, as fallback codecs. Further confirmed by their wiki -->
https://enwiki.peoplefone.com/wiki/Supported_Codecs
peoplefone supports the codecs G711a/u, G722, G729. We also support the
T-38 protocol for fax. (Please contact us, if you want >>to use it.)
Please try to follow the order in the configuration if possible:
g711a
g711u
g722
g729
Sidenote - although you appear to already have it disabled, if you have
problems establishing calls in future make sure you the video >feature
is disabled ("always use video" on the desktop, "Enable video" in the
mobile client).
Video requires RTP/AVPF profiles to be sent in the SDP, which some SIP
proxies don't like. Your invite appears to be OK though (RTP/>AVP),
except for the lack of G.711 codecs advertised in your INVITE.
On Tue, 8 Jan 2019 at 09:04, Andi Soko <[email protected]> wrote:
Hi Sylvain,
Sorry, I've made a wrong assumption. I had an old SIP line from a
different provider in Linphone and didn't know that dailing from the
history also dails the old provider.
So the real issue is apparently:
I cannot dail international numbers with Linphone and peoplephone.at.
Non-international numbers work! With an other App (Zioper) or another
SIP line (localphone.com) everything works.
- Changing the setting "Subsitute + in phone number" doesnt make a
difference
- In the setting "Country code prefix" (if this setting is from
importance) I have tried "+43", "0043" and "0" (country I'm in).
Somehow it doesnt take "" as an entry.
The error is still the same:
Call failed
Cannot call + +4935...9915
Reason was: Not Acceptable Here
[Media Descriptions Syntax error while parsing the SDP]
Find attached the log file.
thanks
Soko
On Tue, 08 Jan 2019 09:43:30 +0100, Sylvain Berfini
<[email protected]> wrote:
Hi Soko,
Can you attach the logs starting from the INVITE sent to the server at
peoplephone.at to the moment you have the media description syntax
error please ?
Thanks.
Le 08/01/2019 à 09:39, Andi Soko a écrit :
Hey guys,
I have linphone installed and using the a SIP line from
peoplephone.at.
I mainly call customers in Germany and when I call their
official/main number "+4935...9926" everything works.
But when I call an extension-number to get to a specific person
(+4935...9915) I get the following error:
Call failed
Cannot call + +4935...9915
Reason was: Not Acceptable Here
[Media Descriptions Syntax error while parsing the SDP]
When I use another SIP client app (i.e. Zioper) with the same SIP
line all numbers are working. But I like linphone interface and would
like to use it :)
can anyone give me some hints were to look at or which option to
change?
thanks
Soko
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