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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