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
[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 220.127.116.11…
Let's say I have two devices configured and the follow call scenarios occur.
 disallow=all allow=g722&ulaw
Polycom phone with g722,ulaw,alaw,g729
 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…
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: file.c:663 ast_openstream_full: File beep does not exist in any format
[May 5 00:44:16] WARNING: file.c:958 ast_streamfile: Unable to open beep (format 0x4 (ulaw)): No such file or directory
[May 5 00:44:16] WARNING: app_record.c:285 record_exec: ast_streamfile…
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:
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: 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…
We need help in enabling g729a codec for our SIP peer that's using Cisco
Our codec is purchased from Digium. We are able to dial out the numbers and answer the call, but there's no
audio. This is when only g729a is allowed. We noticed when they also allow ulaw codec on their side, the codec used
falls back to ulaw and the problem is gone.
I'm getting various codec related warnings after upgrading to 1.8. Did I miss something in the UPGRADE file? Does Asterisk no longer transcode 8-)?
WARNING: channel.c:4909 ast_write: Codec mismatch on channel DAHDI/i1/12124221200-74 setting write format to g722 from ulaw native formats 0x4 (ulaw)
WARNING: channel.c:4909 ast_write: Codec mismatch on channel SIP/interglobe-sip-000001e6 setting write format to g722 from ulaw native formats 0x4 (ulaw)
So I did a little more digging and found a real simple answer:
tells me 'ulaw' or 'siren14' and lets me pick the right file extension
for the record function.
On Tue, Sep 13, 2011 at 5:19 PM, Tom Browning
> Sorry if this is an obvious question and perhaps my Google foo isn't
> right on this one:
> I have calls coming into an Asterisk server that may be using 2
> different codecs. I am recording audio in both cases but the
How can I take advantage of a high-bandwidth CODEC, like G.722, for
internal communications at my site, but use G.711 (alaw/ulaw) for all
other outgoing calls? I need G.711 to support Inband DTMF signaling. As my site has multiple locations that are tied together with IAX
trunks, I was hoping that it would be possible to specify alaw and
ulaw as the first two CODEC choices for the SIP phones, as well as in
their sip.conf configurations, but that I could use the IAX trunks
(with bandwidth=high) to force the phones to…
i have installed asterisk 18.104.22.168 and i have configured 2 account for sip
in order to do internal call when i use x-lite and eyebeam1.5 i can call from 222 to 223 ,and alson from
223 to 222 but when i use my snom 320 i can call from my x-lite or eyebeam1.5 to
snom320 but the issue i can not call from my snom i have this issue just Asterisk 1.8 when i tested with asterisk 1.4 theres
is no problem see the sip.conf and extenssions.conf below and also the cli when i…