Call drop and missing voice

General Help
1

Good morning, I have a PBX with FreePBX 15 in which I have configured trunks with different providers. I am experiencing issues with the Telecom Italia provider. When I make calls and the other party answers, the line immediately drops.

I have conducted various tests, and the UDP ports seem to be correctly forwarded to the PBX from 10000 to 20000. I thought it might be a codec problem, so I added the following parameters to the outgoing trunk configuration:

allow=ulaw&alaw
disallow=all

With some numbers, the issue was resolved, but it stopped working with others. The problem with the others is that when the IVR voice starts, it suddenly becomes completely silent.

Could you help me? Below is the configuration of my trunk.

outgoing

username=+39************
type=friend
secret=******************
qualify=yes
outboundproxy=85.38.249.72
insecure=very
host=telecomitalia.it
fromuser=+39***************
fromdomain=telecomitalia.it
canreinvite=no
allow=ulaw&alaw
disallow=all

incoming

username=+39*************
user=+39*********
type=friend
secret=*******************
insecure=very
host=telecomitalia.it
fromuser=+39************
context=from-trunk&from-pstn-toheader
allow=ulaw&alaw
disallow=all
context=from-trunk

register string

+39*********@telecomitalia.it:*:+39@telecomitalia.it@85.38.249.72:5060

2

Your configuration doesn’t show this, for outbound, and if you did this literally, it would fail, as the disallow all after the allow, means no codecs are available.

If incoming calls work, you are going beyond using an obsolete and unsupported channel driver and using a version of Asterisk that is ancient (unless FreePBX added “very” back in). “very” was deprecated and then removed. It was supposed to make people think about what was really needed, but they didn’t, they simply copied the full form.

This is invalid; there can only be one context.

In this case, I can see no reason for having any inbound settings, but context should be the same for both inbound and outbound, if you need both.

If the first @telecomitalia.it is really needed, you are going to have problems going forwards.

3

might be worth a shot to convert to PJSIP, if that’s a possibility

4

I tried, but the result did not change

5

I tried again, it seems to be a codec problem.
If I give priority to alaw when I call some numbers it works and with others it doesn’t.
When it doesn’t work, I find these errors in the logs

WARNING[12063][C-00000040] chan_sip.c: Asked to transmit frame type alaw, while native formats is (g729) read/write = alaw/alaw
165 [2024-07-30 18:13:58] WARNING[1243][C-00000056] channel.c: Unable to find a codec translation path: (g729) → (alaw)
166 [2024-07-30 18:13:58] WARNING[1243][C-00000056] channel.c: Unable to find a codec translation path: (alaw) → (g729)
[2024-07-30 18:13:58] WARNING[31888][C-00000056] channel.c: No path to translate from SIP/out_-00000077 to SIP/101-00000076
175 [2024-07-30 18:13:58] WARNING[31888][C-00000056] app_dial.c: Had to drop call because I couldn’t make SIP/101-00000076 compatible with SIP/out_***-00000077

reversing the order of the codecs and therefore giving priority to g729 works with some numbers and not with others

the version of freepbx is 15.0.37.5
The g729 codec should be integrated in this version in fact it works if I position it at the top.

in “general sip settings” i have this codec order

2024-07-30_19-27-24
2024-07-30_19-27-24893×528 38 KB

some help?
Thanks

6

Why do you want to use G.729? There are few good reasons to use it these days. I’d suggest the only good reason is to interface with legacy equipment in low wage call centres.

To do anything but pass G.729 through unchanged, Asterisk needs a G.729 codec, and the only officially supported one for FreePBX is a commercial one, with per concurrent half call licensing. Without that, you really need all users to be set for only G.729, and you lose the ability to have conferences, or record calls.

7

if I deactivate g729 many calls are dropped and I can’t hear anything

8

You need to explain your environment. Normally forcing G.729 would be an organisation level policy decision and normal providers would transcode.

Also, how did you install FreePBX? I think the official distribution contains the commercial codec and just requires licensing. As the patents have expired, there are now claimed open source implementations, but if legal, these will have standard C implementations, and will be CPU hogs.

9

disabling g729, most calls start working correctly again, but with some numbers the call starts and then drops.
Below I report the asterisk debug

a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:85.38.249.72:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.202:5060;received=79.9.11.34;branch=z9hG4bK32aaf1b2;rport=5060
From: <sip:+39********[email protected]>;tag=as74001641
To: <sip:********[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:85.38.249.72:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.10.202:5060;received=79.9.11.34;branch=z9hG4bK32aaf1b2;rport=5060
From: <sip:+39********[email protected]>;tag=as74001641
To: <sip:********[email protected]>;tag=p65548t1722438637m578226c112190s1_2133061103-1870899385
Call-ID: [email protected]
CSeq: 103 INVITE
Content-Length: 0
Contact: <sip:[email protected]:5060;transport=udp>
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, MESSAGE, INFO, PUBLISH, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
Accept: application/sdp
Accept: application/isup
Accept: application/xml
Supported: timer
Server: Ericsson MTAS - CXP2010134/1 R30H07

<------------->
--- (14 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:[email protected]:5060;transport=udp>
    -- SIP/out_********975344-0000014c is ringing

<--- Transmitting (NAT) to 10.10.10.101:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.10.101:5060;branch=z9hG4bK662871627;received=10.10.10.101;rport=5060
From: "101" <sip:[email protected]:5060>;tag=662712761
To: <sip:********[email protected]:5060>;tag=as32d22418
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-15.0.37.5(16.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:********[email protected]:5060>
Content-Length: 0


<------------>

<--- SIP read from UDP:85.38.249.72:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.202:5060;received=79.9.11.34;branch=z9hG4bK32aaf1b2;rport=5060
From: <sip:+39********[email protected]>;tag=as74001641
To: <sip:********[email protected]>;tag=p65548t1722438637m578226c112190s1_2133061103-1870899385
Call-ID: [email protected]
CSeq: 103 INVITE
Content-Length: 200
Contact: <sip:[email protected]:5060;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Content-Type: application/sdp
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, MESSAGE, INFO, PUBLISH, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
Accept: application/sdp
Accept: application/isup
Accept: application/xml
Supported: timer, 100rel
P-Asserted-Identity: <sip:0039********[email protected];user=phone>
Session-ID: e6efcee8ea2dd2ae91007c1a26024104
Session-Expires: 360;refresher=uas
Server: Ericsson MTAS - CXP2010134/1 R30H07
Authentication-Info: qop=auth,rspauth="18b57f8bf5fe8e74925efd43794537f0",cnonce="6bffc4b9",nc=00000001

v=0
o=- 861219319772641 2790483040 IN IP4 85.38.249.72
s=IMSS
c=IN IP4 85.38.249.72
t=0 0
m=audio 38342 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=ptime:20
<------------->
--- (19 headers 10 lines) ---
Got SDP version 2790483040 and unique parts [- 861219319772641 IN IP4 85.38.249.72]
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw|g726|g723), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 85.38.249.72:38342
sip_route_dump: route/path hop: <sip:[email protected]:5060;transport=udp>
Transmitting (NAT) to 85.38.249.72:5060:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.10.10.202:5060;branch=z9hG4bK50f67258;rport
Max-Forwards: 70
From: <sip:+39********[email protected]>;tag=as74001641
To: <sip:********[email protected]>;tag=p65548t1722438637m578226c112190s1_2133061103-1870899385
Contact: <sip:+39********[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: FPBX-15.0.37.5(16.15.0)
Content-Length: 0


---
    -- SIP/out_********975344-0000014c answered SIP/101-0000014b
    -- SIP/out_********975344-0000014c Internal Gosub(sub-send-obroute-email,s,1(********6944,********6944,5,1722438637,,********975344)) start
    -- Executing [s@sub-send-obroute-email:1] GotoIf("SIP/out_********975344-0000014c", "0?sendEmail") in new stack
    -- Executing [s@sub-send-obroute-email:2] NoOp("SIP/out_********975344-0000014c", "email notifications disabled..exiting.") in new stack
    -- Executing [s@sub-send-obroute-email:3] Return("SIP/out_********975344-0000014c", "") in new stack
  == Spawn extension (from-trunk-sip-out_********975344, , 1) exited non-zero on 'SIP/out_********975344-0000014c'
    -- SIP/out_********975344-0000014c Internal Gosub(sub-send-obroute-email,s,1(********6944,********6944,5,1722438637,,********975344)) complete GOSUB_RETVAL=
Audio is at 16024
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 10.10.10.101:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.101:5060;branch=z9hG4bK662871627;received=10.10.10.101;rport=5060
From: "101" <sip:[email protected]:5060>;tag=662712761
To: <sip:********[email protected]:5060>;tag=as32d22418
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-15.0.37.5(16.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:********[email protected]:5060>
P-Asserted-Identity: "********6944" <sip:********[email protected]>
Content-Type: application/sdp
Content-Length: 272

v=0
o=root 87308553 87308553 IN IP4 10.10.10.202
s=Asterisk PBX 16.15.0
c=IN IP4 10.10.10.202
t=0 0
m=audio 16024 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>

<--- SIP read from UDP:10.10.10.101:5060 --->
ACK sip:********[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.101:5060;branch=z9hG4bK662967576
From: "101" <sip:[email protected]:5060>;tag=662712761
To: <sip:********[email protected]:5060>;tag=as32d22418
Call-ID: [email protected]
CSeq: 2 ACK
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
User-Agent: Yealink SIP-T54W 96.86.0.53
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
    -- Channel SIP/out_********975344-0000014c joined 'simple_bridge' basic-bridge <3debc7e2-5859-4fa5-9fbd-b9e32ba31bb3>
    -- Channel SIP/101-0000014b joined 'simple_bridge' basic-bridge <3debc7e2-5859-4fa5-9fbd-b9e32ba31bb3>

<--- SIP read from UDP:85.38.249.72:5060 --->
INVITE sip:+39********[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 85.38.249.72:5060;branch=z9hG4bKc7ec5a0038a8svbd5ab0sb00000k1.1
To: <sip:+39********[email protected];user=phone>;tag=as74001641
From: <sip:+39********[email protected];user=phone>;tag=p65548t1722438637m578226c112190s1_2133061103-1870899385
Call-ID: [email protected]
CSeq: 104 INVITE
Max-Forwards: 66
Content-Length: 237
Contact: <sip:[email protected]:5060;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Content-Type: application/sdp
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, MESSAGE, INFO, PUBLISH, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
Supported: timer,100rel
Min-SE: 180
Session-Expires: 360;refresher=uac

v=0
o=- 861219319772641 2790483041 IN IP4 85.38.249.72
s=IMSS
c=IN IP4 85.38.249.72
t=0 0
m=audio 38342 RTP/AVP 18 101
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=silenceSupp:off - - - -
<------------->
--- (14 headers 11 lines) ---
Sending to 85.38.249.72:5060 (NAT)
Comparing SDP version 2790483040 -> 2790483041 and unique parts [- 861219319772641 IN IP4 85.38.249.72] -> [- 861219319772641 IN IP4 85.38.249.72]
Found RTP audio format 18
Found RTP audio format 101
Found audio description format telephone-event for ID 101
[2024-07-31 17:10:40] NOTICE[1243][C-000000bf]: chan_sip.c:10973 process_sdp: No compatible codecs, not accepting this offer!

<--- Reliably Transmitting (NAT) to 85.38.249.72:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 85.38.249.72:5060;branch=z9hG4bKc7ec5a0038a8svbd5ab0sb00000k1.1;received=85.38.249.72;rport=5060
From: <sip:+39********[email protected];user=phone>;tag=p65548t1722438637m578226c112190s1_2133061103-1870899385
To: <sip:+39********[email protected];user=phone>;tag=as74001641
Call-ID: [email protected]
CSeq: 104 INVITE
Server: FPBX-15.0.37.5(16.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 360;refresher=uac
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<------------>

<--- SIP read from UDP:85.38.249.72:5060 --->
ACK sip:+39********[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 85.38.249.72:5060;branch=z9hG4bKc7ec5a0038a8svbd5ab0sb00000k1.1
CSeq: 104 ACK
To: <sip:+39********[email protected];user=phone>;tag=as74001641
From: <sip:+39********[email protected];user=phone>;tag=p65548t1722438637m578226c112190s1_2133061103-1870899385
Call-ID: [email protected]
Max-Forwards: 66
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
raspbx*CLI> sip set debug off
SIP Debugging Disabled
raspbx*CLI>

I have FreePBX 15.0.37.5 installed on raspberry PI4 with 2 trunks

The first Ehiweb where there is no problem by disabling g729
The second Telecom Italia where the problem just described occurs
My Asterisk server has the following codecs

      ID TYPE  NAME         FORMAT           DESCRIPTION
--------------------------------------------------------------------------------                       ----------------
      31 image png          png              (PNG Image)
       6 audio g726         g726             (G.726 RFC3551)
       4 audio alaw         alaw             (G.711 a-law)
       2 audio g723         g723             (G.723.1)
      20 audio speex        speex            (SpeeX)
      21 audio speex        speex16          (SpeeX 16khz)
      22 audio speex        speex32          (SpeeX 32khz)
      24 audio g722         g722             (G722)
      25 audio siren7       siren7           (ITU G.722.1 (Siren7, licensed from                        Polycom))
      32 video h261         h261             (H.261 video)
      33 video h263         h263             (H.263 video)
       8 audio adpcm        adpcm            (Dialogic ADPCM)
      36 video h265         h265             (H.265 video)
      44 audio silk         silk8            (SILK Codec (8 KHz))
      45 audio silk         silk12           (SILK Codec (12 KHz))
      46 audio silk         silk16           (SILK Codec (16 KHz))
      47 audio silk         silk24           (SILK Codec (24 KHz))
      28 audio g719         g719             (ITU G.719)
      34 video h263p        h263p            (H.263+ video)
      35 video h264         h264             (H.264 video)
      19 audio g729         g729             (G.729A)
       9 audio slin         slin             (16 bit Signed Linear PCM)
      10 audio slin         slin12           (16 bit Signed Linear PCM (12kHz))
      11 audio slin         slin16           (16 bit Signed Linear PCM (16kHz))
      12 audio slin         slin24           (16 bit Signed Linear PCM (24kHz))
      13 audio slin         slin32           (16 bit Signed Linear PCM (32kHz))
      14 audio slin         slin44           (16 bit Signed Linear PCM (44kHz))
      15 audio slin         slin48           (16 bit Signed Linear PCM (48kHz))
      16 audio slin         slin96           (16 bit Signed Linear PCM (96kHz))
      17 audio slin         slin192          (16 bit Signed Linear PCM (192kHz))
       3 audio ulaw         ulaw             (G.711 u-law)
      18 audio lpc10        lpc10            (LPC10)
      27 audio testlaw      testlaw          (G.711 test-law)
      43 audio none         none             (<Null> codec)
      42 image t38          t38              (T.38 UDPTL Fax)
      39 video vp9          vp9              (VP9 video)
      38 video vp8          vp8              (VP8 video)
       5 audio gsm          gsm              (GSM)
      37 video mpeg4        mpeg4            (MPEG4 video)
      23 audio ilbc         ilbc             (iLBC)
      40 text  red          red              (T.140 Realtime Text with redundanc                       y)
      41 text  t140         t140             (Passthrough T.140 Realtime Text)
      29 audio opus         opus             (Opus Codec)
      30 image jpeg         jpeg             (JPEG image)
       7 audio g726aal2     g726aal2         (G.726 AAL2)
       1 audio codec2       codec2           (Codec 2)
      26 audio siren14      siren14          (ITU G.722.1 Annex C, (Siren14, lic                       ensed from Polycom))

Do you need any other details?

10

IMSS has tried to force a renegotiation of the codec to G.729, even though another one of its initial offers has been accepted. That seems broken. My guess is that they don’t have G.729 licences either, or have run out, so are trying to pass the buck to you, to do the transcoding.

Your list of codecs represents what is known, but doesn’t meant that Asterisk can actually transcode in or out of the codec. To see if it has the software to do that, you would need to run “core show translation” and see if there was a finiite cost to translate between G.729 and the codec used by the other side of the call. To record, or do conferences, you need a translation path to slin. I’m not sure how that handles the case where the codec is installed, but you have too few licences.

As I said there is, I believe, now an open source codec that can be legal, but you need to be careful to only use the version that uses the pure C implementation, as the version that uses multiple data single instruction processing to speed it up on Intel machines is using a library that is only licensed for evaluation purposes, and is therefore not GPL compatible, and cannot be used in a production environment.

However, there are better low bit rate codecs than G.729 and they were never patent encumbered, so there is no technical reason to use G.729. I guess you have legacy equipment, or equipment that has been misconfigured.

11

thank you so much for the support.
I would like to understand this point better (sorry I’m Italian and I don’t understand much English)

I activate the g729 only because some calls require it and if they don’t find it enabled they are interrupted. How can I get around the problem and not activate the g729 codec?
Thanks for your patience

12 
 Translation times between formats (in microseconds) for one second of d                                                             ata
          Source Format (Rows) Destination Format (Columns)

          codec2  ulaw  alaw   gsm  g726 g726aal2 adpcm slin8 slin12 slin16 slin                                                             24 slin32 slin44 slin48 slin96 slin192 lpc10 speex8 speex16 speex32  ilbc  g722                                                              testlaw  opus
   codec2      - 15000 15000 15000 15000    15000 15000  9000  17000  17000  170                                                             00  17000  17000  17000  17000   17000 15000  15000   23000   23000 15000 17250                                                                15000 15000
     ulaw  15000     -  9150 15000 15000    15000 15000  9000  17000  17000  170                                                             00  17000  17000  17000  17000   17000 15000  15000   23000   23000 15000 17250                                                                15000 15000
     alaw  15000  9150     - 15000 15000    15000 15000  9000  17000  17000  170                                                             00  17000  17000  17000  17000   17000 15000  15000   23000   23000 15000 17250                                                                15000 15000
      gsm  15000 15000 15000     - 15000    15000 15000  9000  17000  17000  170                                                             00  17000  17000  17000  17000   17000 15000  15000   23000   23000 15000 17250                                                                15000 15000
     g726  15000 15000 15000 15000     -    15000 15000  9000  17000  17000  170                                                             00  17000  17000  17000  17000   17000 15000  15000   23000   23000 15000 17250                                                                15000 15000
 g726aal2  15000 15000 15000 15000 15000        - 15000  9000  17000  17000  170                                                             00  17000  17000  17000  17000   17000 15000  15000   23000   23000 15000 17250                                                                15000 15000
    adpcm  15000 15000 15000 15000 15000    15000     -  9000  17000  17000  170                                                             00  17000  17000  17000  17000   17000 15000  15000   23000   23000 15000 17250                                                                15000 15000
    slin8   6000  6000  6000  6000  6000     6000  6000     -   8000   8000   80                                                             00   8000   8000   8000   8000    8000  6000   6000   14000   14000  6000  8250                                                                 6000  6000
   slin12  14500 14500 14500 14500 14500    14500 14500  8500      -   8000   80                                                             00   8000   8000   8000   8000    8000 14500  14500   14000   14000 14500 14000                                                                14500  5999
   slin16  14500 14500 14500 14500 14500    14500 14500  8500   8500      -   80                                                             00   8000   8000   8000   8000    8000 14500  14500    6000   14000 14500  6000                                                                14500  5999
   slin24  14500 14500 14500 14500 14500    14500 14500  8500   8500   8500                                                                   -   8000   8000   8000   8000    8000 14500  14500   14500   14000 14500 14500                                                                14500  5999
   slin32  14500 14500 14500 14500 14500    14500 14500  8500   8500   8500   85                                                             00      -   8000   8000   8000    8000 14500  14500   14500    6000 14500 14500                                                                14500 13999
   slin44  14500 14500 14500 14500 14500    14500 14500  8500   8500   8500   85                                                             00   8500      -   8000   8000    8000 14500  14500   14500   14500 14500 14500                                                                14500 13999
   slin48  14500 14500 14500 14500 14500    14500 14500  8500   8500   8500   85                                                             00   8500   8500      -   8000    8000 14500  14500   14500   14500 14500 14500                                                                14500  5999
   slin96  14500 14500 14500 14500 14500    14500 14500  8500   8500   8500   85                                                             00   8500   8500   8500      -    8000 14500  14500   14500   14500 14500 14500                                                                14500 14499
  slin192  14500 14500 14500 14500 14500    14500 14500  8500   8500   8500   85                                                             00   8500   8500   8500   8500       - 14500  14500   14500   14500 14500 14500                                                                14500 14499
    lpc10  15000 15000 15000 15000 15000    15000 15000  9000  17000  17000  170                                                             00  17000  17000  17000  17000   17000     -  15000   23000   23000 15000 17250                                                                15000 15000
   speex8  15000 15000 15000 15000 15000    15000 15000  9000  17000  17000  170                                                             00  17000  17000  17000  17000   17000 15000      -   23000   23000 15000 17250                                                                15000 15000
  speex16  23500 23500 23500 23500 23500    23500 23500 17500  17500   9000  170                                                             00  17000  17000  17000  17000   17000 23500  23500       -   23000 23500 15000                                                                23500 14999
  speex32  23500 23500 23500 23500 23500    23500 23500 17500  17500  17500  175                                                             00   9000  17000  17000  17000   17000 23500  23500   23500       - 23500 23500                                                                23500 22999
     ilbc  15000 15000 15000 15000 15000    15000 15000  9000  17000  17000  170                                                             00  17000  17000  17000  17000   17000 15000  15000   23000   23000     - 17250                                                                15000 15000
     g722  15600 15600 15600 15600 15600    15600 15600  9600  17500   9000  170                                                             00  17000  17000  17000  17000   17000 15600  15600   15000   23000 15600     -                                                                15600 14999
  testlaw  15000 15000 15000 15000 15000    15000 15000  9000  17000  17000  170                                                             00  17000  17000  17000  17000   17000 15000  15000   23000   23000 15000 17250                                                                    - 15000
     opus  15000 15000 15000 15000 15000    15000 15000  9000   8999   8999   89                                                             99  16999  16999   8999  16999   16999 15000  15000   14999   22999 15000 14999                                                                15000     -
13

Forego call recording, conferences, and anything else that requires Asterisk to make internal use of the media stream. Set all trunks and extensions to ONLY use G.729.

14

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