Forcing GSM On Certain Extensions

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I'm trying to force GSM when I call on certain extension but I'm getting connected with "ulaw" Which is not suitable when bandwidth is low and slow.

my phone is iax-322

in iax.conf

[iaxy-322] ... disallow=all allow=gsm allow=ulaw allow=alaw

[zoiper_kathy_old_phone] ... disallow=all allow=gsm allow=ilbc allow=ulaw allow=alaw allow=speex

I've define "allow=gsm" in as the first codec so it should take priority, isn't it? but when I call I get connected with "ulaw"

Incoming call is getting IN as GSM from that phone but outgoing is going out as "ulaw" why?

In "zoiper" the GSM is enable as well and as first codec.

Called IAX2/zoiper_kathy_old_phone -- Call accepted by 66.18.210.217…

Asterisk Users 6 months ago 2 Answer

Why Doesn't Asterisk Try To Prevent Transcoding

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Let's say I have two devices configured and the follow call scenarios occur.

[100] disallow=all allow=g722&ulaw

Polycom phone with g722,ulaw,alaw,g729

[101] disallow=all allow=ulaw

Polycom phone with g722,ulaw,alaw,g729

101 dials 100 -> ulaw to ulaw is chosen 100 dials 101 -> g722 to ulaw is chosen

Ideally when 100 dials 101 ulaw would be chosen since it is the common format. Looking into this deeper

Device 100 sends INVITE to Asterisk offering g722,ulaw,alaw,g729 Asterisk sends INVITE to device 101 offering ulaw Device 101 sends 200 OK to Asterisk offering ulaw Asterisk sends 200 OK to device 100 offering g722,ulaw

I can prevent transcoding by adding SIP_CODEC_INBOUND=ulaw to the dialplan…

Asterisk Users 1.6 years ago 12 Answer

Sound file format and Asterisk 1.8.11-cert1

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Hi All; I installed Asterisk 1.8.11-cert1, and it look like the default is ulaw for the sound files. How I can fix this? Athough file beep.gsm is existed under path (/var/lib/asterisk/sounds/en), but when I used the Record function, it gave me the following (so I am sure there is something that can let asterisk accept beep.gsm), what could be? [May 5 00:44:16] WARNING[2262]: file.c:663 ast_openstream_full: File beep does not exist in any format
[May 5 00:44:16] WARNING[2262]: file.c:958 ast_streamfile: Unable to open beep (format 0x4 (ulaw)): No such file or directory
[May 5 00:44:16] WARNING[2262]: app_record.c:285 record_exec: ast_streamfile…

Asterisk Users 3.2 years ago 0 Answer

Same provider - IAX sounds bad, SIP sounds great

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On my Asterisk system, I'm using a provider that provides both IAX2 and
SIP connectivity. Personally, I'd prefer to use IAX2, and that's what my account is setup
to use. However, I'm having a problem: With IAX2:
- Incoming Voice from my Provider -> Asterisk = Sounds great
- Outgoing Voice from Asterisk -> my Provider = Sounds terrible By "terrible," I mean skips, stutters, and distortion. It can be
difficult (sometimes impossible) to understand. It doesn't matter what
codec I use (at least between G.729, GSM, or ulaw). On the other hand:

Asterisk Users 3.4 years ago 46 Answer

Dropping incompatible voice frame error

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Greetings, We have an old analog phone system running Asterisk 1.2.13 (not my choice lol). Everything has been working wonderfully until today. The site is experiencing dropped and missed calls. When I tried calling the site, I did get through however the CLI was flooded with hundreds of copies of the following:
Jan 25 15:46:54 NOTICE[2343]: channel.c:1956 ast_read: Dropping incompatible voice frame on Local/4227@from-sip-a3d8,2 of format ulaw since our native format has changed to slin I tried Googling the error but unfortunately, there were lots of reports of the problem and not a single solution or fix. My thinking…

Asterisk Users 3.5 years ago 0 Answer