Calls droped due to codecs problem

gelato_33 · January 24, 2022, 2:50pm

Hello community,

Can anyone explain why this call drops?


    -- Executing [5555hotel:1] AGI("SIP/igh41432-000004e1", "callerid.php") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/callerid.php
 callerid.php: Tenim agi_request=callerid.php
 callerid.php: Tenim agi_channel=SIP/igh41432-000004e1
 callerid.php: Tenim agi_language=es
 callerid.php: Tenim agi_type=SIP
 callerid.php: Tenim agi_uniqueid=1643034155.1481
 callerid.php: Tenim agi_version=certified/11.6-cert12
 callerid.php: Tenim agi_callerid=4143
 callerid.php: Tenim agi_calleridname=Sistemas
 callerid.php: Tenim agi_callingpres=0
 callerid.php: Tenim agi_callingani2=0
 callerid.php: Tenim agi_callington=0
 callerid.php: Tenim agi_callingtns=0
 callerid.php: Tenim agi_dnid=5555
 callerid.php: Tenim agi_rdnis=unknown
 callerid.php: Tenim agi_context=hotel
 callerid.php: Tenim agi_extension=5555
 callerid.php: Tenim agi_priority=1
 callerid.php: Tenim agi_enhanced=0.0
 callerid.php: Tenim agi_accountcode=
 callerid.php: Tenim agi_threadid=139690683664128
 callerid.php: SELECT * from habitaciones where habitacion = '4143'
    -- <SIP/igh41432-000004e1>AGI Script callerid.php completed, returning 0
    -- Executing [5555hotel:2] Dial("SIP/igh41432-000004e1", "SIP/,60,tTr") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 13058
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to :7065:
INVITE sip::7065 SIP/2.0
Via: SIP/2.0/UDP :7065;branch=;rport
Max-Forwards: 70
From: "Sistemas" <sip::7065>;tag=as289f2fca
To: <sip::7065>
Contact: <sip:i:7065>
Call-ID: :7065
CSeq: 102 INVITE
User-Agent: Asterisk PBX certified/11.6-cert12
Date: Mon, 24 Jan 2022 14:22:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Sistemas" <>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 273

v=0
o=root 1200536259 1200536259 IN IP4 
s=Asterisk PBX certified/11.6-cert12
c=IN IP4 
t=0 0
m=audio 13058 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/ic

<--- SIP read from UDP::7065 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP :7065;branch=z9hG4b;rport=7065
From: "Sistemas" <sip:i@:7065>;tag=as289f2fca
To: <sip::7065>;tag=as16eb3594
Call-ID: 7fc:7065
CSeq: 102 INVITE
Server: Asterisk PBX certified/13.18-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1dfadde1"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to :7065:
ACK sip:6:7065 SIP/2.0
Via: SIP/2.0/UDP :7065;branch=z9hG4bK33f5d41e;rport
Max-Forwards: 70
From: "Sistemas" <sip:i:7065>;tag=as289f2fca
To: <sip:67065>;tag=as16eb3594
Contact: <sip:i:7065>
Call-ID: 7
CSeq: 102 ACK
User-Agent: Asterisk PBX certified/11.6-cert12
Content-Length: 0


---
Audio is at 13058
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to :7065:
INVITE sip::7065 SIP/2.0
Via: SIP/2.0/UDP 3c;rport
Max-Forwards: 70
From: "Sistemas" <sip::7065>;tag=as289f2fca
To: <sip:7065>
Contact: <sip:>
Call-ID: 7fcc114b2db7065
CSeq: 103 INVITE
User-Agent: Asterisk PBX certified/11.6-cert12
Authorization: Digest username="", realm="asterisk", algorithm=MD5, uri="sip::7065", nonce="1dfadde1", response="c70d05148a221806"
Date: Mon, 24 Jan 2022 14:22:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Sistemas" <sip:4143>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 273

v=0
o=root 1200536259 1200536260 IN IP4 
s=Asterisk PBX certified/11.6-cert12
c=IN IP4 
t=0 0
m=audio 13058 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP::7065 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 7065;branch=z9bKb361c;r;rport=7065
From: "Sistemas" <sip::7065>;tag=as289f2fca
To: <sip:7065>
Call-ID: 7fcc
CSeq: 103 INVITE
Server: Asterisk PBX certified/13.18-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:6:7065>
Content-Length: 0

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

<--- SIP read from UDP::7065 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 7065;branch=z9hGc;;rport=7065
From: "Sistemas" <sip:i1:7065>;tag=as289f2fca
To: <sip:67065>;tag=as29a93d14
Call-ID: 7fcc114
CSeq: 103 INVITE
Server: Asterisk PBX certified/13.18-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:>
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
list_route: hop: <sip:67065>
    -- SIP/voipminic-000004e2 is ringing

<--- SIP read from UDP::7065 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP rport=7065
From: "Sistemas" <sip::7065>;tag=as289f2fca
To: <sip:;tag=as29a93d14
Call-ID: 7fc
CSeq: 103 INVITE
Server: Asterisk PBX certified/13.18-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:7065>
Content-Type: application/sdp
Require: timer
Content-Length: 293

v=0
o=root 1637700665 1637700665 IN IP4 
s=Asterisk PBX certified/13.18-cert2
c=IN IP4 
t=0 0
m=audio 13144 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:230
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (g729), peer - audio=(g729)/video=(nothing)/text=(nothing), combined - (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 
list_route: hop: <sip:6:7065>
set_destination: Parsing <sip:6:7065> for address/port to send to
set_destination: set destination to 17065
Transmitting (NAT) to 5:
ACK sip::7065 SIP/2.0
Via: SIP/2.0/UDP7065;86;rport
Max-Forwards: 70
From: "Sistemas" <sip:065>;tag=as289f2fca
To: <sip:6:7065>;tag=as29a93d14
Contact: <sip:i:7065>
Call-ID: 7fcc114b2db
CSeq: 103 ACK
User-Agent: Asterisk PBX certified/11.6-cert12
Content-Length: 0


---
    -- SIP/voipminic-000004e2 answered SIP/igh41432-000004e1
[Jan 24 15:22:41] WARNING[20258][C-0000294c]: channel.c:6165 ast_channel_make_compatible_helper: No path to translate from SIP/igh41432-000004e1 to SIP/voipminic-000004e2
[Jan 24 15:22:41] WARNING[20258][C-0000294c]: app_dial.c:3032 dial_exec_full: Had to drop call because I couldn't make SIP/igh41432-000004e1 compatible with SIP/voipminic-000004e2
Scheduling destruction of SIP dialog '7fcc114b2dbe48e43406004123239764@3.121.58.251:7065' in 6400 ms (Method: INVITE)

Thank you,

If u need more info or output, please tell me!

jcolp · January 24, 2022, 2:51pm

One side is G729, the other is likely not G729. Your Asterisk can’t transcode, and the call drops.

gelato_33 · January 24, 2022, 3:01pm

Hi jcolp,

Sure this is the problem, but:

I can change the config in order to make this call works if I put only allow g729 in sip.conf but It makes another provider to fail if I restrict alaw. I tried allowing both (alaw and g729) but first codec is always prefered and, depending on the trunk (peer-provider or whatever name u like) the call goes well or Asterisk try to transcode G729 but it can’t bc i have no license.

I surely can avoid g729 but one provider is offering me just g729. Maybe I cant talk to them to change this, BUT, and this is what is weird to me, this problems showed up when we migrate asterisk from local to aws.

Maybe all of this is confusing…

Another thing, what does this line mean anyway?

Capabilities: us - (g729), peer - audio=(g729)/video=(nothing)/text=(nothing), combined - (g729)

Thank you,

david551 · January 24, 2022, 3:01pm

One would have needed the logging for the incoming leg.

However, Asterisk 11 was end of life over four years ago and security fixes only over five years ago.

I also note you are using a certified version. The only valid reason for doing that would be that you had a support contract with Sangoma, but I’m certain that such a support contract wouldn’t cover Asterisk 11.

jcolp · January 24, 2022, 3:04pm

It means what it says. You’ve configured g729, they’ve offered g729, together that means the call is g729.

gelato_33 · January 24, 2022, 3:09pm

Hi David, thank you for answering

I’ll try to answer your questions,

I dont understand what are you asking in your first question: ¿Which is the incoming leg here? It’s and outbound call from a sip phone to a mobile using one of my providers, if I change my config i can make this call work, so is not a problem of capabilities but a configuration one I think.

Ok, is end of life, maybe I need to update as soon as possible. Thanks

Regarding my certified version, I dont know, we payed another company to install and make the initial configuration. ¿anything i should know about this?

gelato_33 · January 24, 2022, 3:24pm

Hi again.

Ok, same call, different configuration, calls goes out.

 == Using SIP RTP CoS mark 5
    -- Executing [5555@pruebas:1] Dial("SIP/igh41432-000004f7", "SIP/Tm") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 13774
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to7065:
INVITE sip:@7065 SIP/2.0
Via: SIP/2.0/UDP7065;branch=z9hG4bK292dc9cc;rport
Max-Forwards: 70
From: "Sistemas" <sip:i7065>;tag=as6d42a52f
To: <sip::7065>
Contact: <sip:7065>
Call-ID: 4:7065
CSeq: 102 INVITE
User-Agent: Asterisk PBX certified/11.6-cert12
Date: Mon, 24 Jan 2022 15:11:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Sistemas" <s>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 271

v=0
o=root 882067297 882067297 IN IP4 
s=Asterisk PBX certified/11.6-cert12
c=IN IP4
t=0 0
m=audio 13774 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP@voipminic
    -- Started music on hold, class 'default', on SIP/igh41432-000004f7

<--- SIP read from UDP:065 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 3rport=7065
From: "Sistemas" <sip::7065>;tag=as6d42a52f
To: <sip:@>;tag=as4e7cc36d
Call-ID: 41d37d1072257d2a007e5738307adcf2@65
CSeq: 102 INVITE
Server: Asterisk PBX certified/13.18-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3076e978"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to :
ACK sip:67065 SIP/2.0
Via: SIP/2.0/UDP 5;branch=z9hG4bK292dc9cc;rport
Max-Forwards: 70
From: "Sistemas" <sip:i7065>;tag=as6d42a52f
To: <sip:6:7065>;tag=as4e7cc36d
Contact: <7065>
Call-ID: 41d37d1072257d2a007e5738307adcf2@7065
CSeq: 102 ACK
User-Agent: Asterisk PBX certified/11.6-cert12
Content-Length: 0


---
Audio is at 13774
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to:
INVITE sip::7065 SIP/2.0
Via: SIP/2.0/UDP;branch=z9hG4bK6d70d3f5;rport
Max-Forwards: 70
From: "Sistemas" <sip:i:7065>;tag=as6d42a52f
To: <sip:7065>
Contact: <7065>
Call-ID: 41d37d1072257d2a007e5738307adcf
CSeq: 103 INVITE
User-Agent: Asterisk PBX certified/11.6-cert12
Authorization: Digest username="ibizagranhotel", realm="asterisk", algorithm=MD5, uri="sip:", nonce="3076e978", response="7964847c5bfe1195249332416c2343e6"
Date: Mon, 24 Jan 2022 15:11:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Sistemas" <sip>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 271

v=0
o=root 882067297 882067298 IN IP4 
s=Asterisk PBX certified/11.6-cert12
c=IN IP4 
t=0 0
m=audio 13774 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:1065 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ;rport=7065
From: "Sistemas" 7065>;tag=as6d42a52f
To: <sip7065>
Call-ID: 41d37d1072:7065
CSeq: 103 INVITE
Server: Asterisk PBX certified/13.18-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:7065>
Content-Length: 0

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

<--- SIP read from UDP::7065 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ;branch=z9hG4bK6d;rport=7065
From: "Sistemas" <sip::7065>;tag=as6d42a52f
To: <sip::7065>;tag=as05f38f2e
Call-ID: 41d37d1072257d2a007e5738307adcf2:7065
CSeq: 103 INVITE
Server: Asterisk PBX certified/13.18-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip::7065>
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
list_route: hop: <sip::7065>
    -- SIP/-000004f8 is ringing
       > 0x7f0c7470e920 -- Probation passed - setting RTP source address to :5104
       > 0x7f0c7470e920 -- Probation passed - setting RTP source address to:5104
[Jan 24 16:11:13] NOTICE[1213]: chan_sip.c:27695 handle_request_subscribe: Received SIP subscribe for peer without mailbox: hab2517
[Jan 24 16:11:14] NOTICE[1213]: chan_sip.c:27695 handle_request_subscribe: Received SIP subscribe for peer without mailbox: hab3102
[Jan 24 16:11:15] NOTICE[1213]: chan_sip.c:27695 handle_request_subscribe: Received SIP subscribe for peer without mailbox: hab3106

<--- SIP read from UDP::7065 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 7065;branch=z9hG4bK6d70d3f5;received=;rport=7065
From: "Sistemas" <sip:i:7065>;tag=as6d42a52f
To: <sip:com:7065>;tag=as05f38f2e
Call-ID: 4165
CSeq: 103 INVITE
Server: Asterisk PBX certified/13.18-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:7065>
Content-Type: application/sdp
Require: timer
Content-Length: 289

v=0
o=root 58316748 58316748 IN IP4 
s=Asterisk PBX certified/13.18-cert2
c=IN IP4 
t=0 0
m=audio 11668 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:230
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (g729), peer - audio=(g729)/video=(nothing)/text=(nothing), combined - (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 11668
list_route: hop: <sip::7065>
set_destination: Parsing <sip::7065> for address/port to send to
set_destination: set destination to :7065
Transmitting (NAT) to :7065:
ACK sip::7065 SIP/2.0
Via: SIP/2.0/UDP :7065;branch=z9hG4bK1a6565de;rport
Max-Forwards: 70
From: "Sistemas" <sip:7065>;tag=as6d42a52f
To: <sip:7065>;tag=as05f38f2e
Contact: <sip:7065>
Call-ID: 41d37d1072257d2a007e5738307adcf2@:7065
CSeq: 103 ACK
User-Agent: Asterisk PBX certified/11.6-cert12
Content-Length: 0


---
    -- SIP/voipminic-000004f8 answered SIP/igh41432-000004f7
    -- Stopped music on hold on SIP/igh41432-000004f7
       > 0x7f0c6c02b580 -- Probation passed - setting RTP source address to :11668

<--- SIP read from UDP:7065 --->
BYE sip:ibizagranhotel@:7065 SIP/2.0
Via: SIP/2.0/UDP :7065;branch=z9hG4bK3d4c97bc;rport
Max-Forwards: 70
From: <sip:@:7065>;tag=as05f38f2e
To: "Sistemas" <sip:l@:7065>;tag=as6d42a52f
Call-ID: 41d37d:7065
CSeq: 102 BYE
User-Agent: Asterisk PBX certified/13.18-cert2
Proxy-Authorization: Digest username="", realm="asterisk", algorithm=MD5, uri="sip:pbx.voipminic.com", nonce="3076e978", response="f75d53cbb792aa8f3ebf3308f06bfd3e"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 7065 (NAT)
Scheduling destruction of SIP dialog '41d37d1072257d2a007e5738307adcf2@7065' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to7065 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 7065;branch=z9hG4bK3d4c97bc;received=1;rport=7065
From: <sip:@pbx.voipminic.com:7065>;tag=as05f38f2e
To: "Sistemas" <sip::7065>;tag=as6d42a52f
Call-ID: 41d37d10722065
CSeq: 102 BYE
Server: Asterisk PBX certified/11.6-cert12
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

So the problem here is that im not able to configure correctly two differents trunks if one of them want to use only alaw and the other one only g729

Is there a way to solve this without installing g729?

Thank you,

david551 · January 24, 2022, 3:40pm

Again, you are not providing the logging from the incoming leg (the call from SIP/igh41432 Also not that your logs are garbled, because you didn’t mark them up with the </> button.

The only reason the call could have worked is because the incoming leg did allow G.729 or you do have at least one G729 licence.

gelato_33 · January 24, 2022, 7:27pm

Sure I Don’t Have any license.

So, assuming what you say "the incoming leg is allowing g729, and it is since call is succesful, why if I change this in the internal context:

disallow=All
allow=g729

(call is ok)

For this…

Disallow=All
Allow=alaw
Allow=g729

The call drops because Asterisk need to transcode?

sedwards · January 24, 2022, 8:16pm

If you don’t have g729 (licensed or otherwise) and you’re doing things that require Asterisk to transcode, why are you surprised Asterisk can’t transcode?

The only way you can process a ‘g729’ call is if every endpoint involved has g729 and every file you play is encoded as g729 and you don’t do anything that requires transcoding – like conferencing.

(Just my personal experience… Every time I’ve tried g729, people (who are used to xlaw) complain. So I don’t bother with g729 anymore.)

david551 · January 24, 2022, 8:19pm

If used on the incoming side, that allows them to use either G.729 or G.711 A-Law, with a preference for G.711. I think there is now an option, when Asterisk is responding, as to whether it send all of these, or just the caller’s first choice. But, in any case, if they actually choose to send G.711, Asterisk will need to transcode to get it to the side that only supports G.729, and, to do that, it needs both the g729 codec module, and sufficient licences.

If both sides use G.729, Asterisk can pass this through without understanding how to interpret the contents.

gelato_33 · January 25, 2022, 10:21am

Thank you for trying to help me, ok, maybe I need to reformulate my issue and give you a general approach.

I never had problems with codecs before when I have Asterisk in my LAN. The issue starts when we migrate asterisk to aws.

I have two Epigy QXISDN4 to connect to two PSTN lines and one connection with another asterisk that provides us a SIP trunk.

In debug I can see that peers of PSTN lines offer always only alaw, SIP peer offers only g729.

My config regarding codecs in peers and contexts was always the same:

diasallow=all
allow=g729
allow=allaw

With this, everything was working like a charm without any g729 license.

After migration to aws, none of them was working due to codecs missmatches or the need of transcoding. ¿any idea why this could happen?

So, my question is, is there a way to use alaw when I call to PSTN peers and g729 when I talk to the sip provider? Same thing, how can I make a phone that belongs to a certain context to use one or another codecs according to the peer I need to use?

I don’t know if it has any sense to you,

Thank you

gelato_33 · January 25, 2022, 3:12pm

Oh Lord, my apologies to the people in this thread.

Finally, I could talk with the one who installed and configured our Asterisk.

It HAD a digium g.729 license, but the license WAS LOST in the migration to AWS, this explains the behaviour I tried to share here.

Thank you,

system · February 24, 2022, 3:13pm

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Calls droped due to codecs problem

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