[asterisk-bugs] [JIRA] (ASTERISK-24320) Incoming audio not working

Thu Sep 11 18:38:28 CDT 2014

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24320?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-24320:
------------------------------------

    Description: 
Hello
We are using asterisk 1.4.10.1 with digium gateway G100

Due to problem with our old redfone T1 gateway we replaced with digium g100 gateway.

After changing the digium G100 Outgoing is working fine but no audio for incoming calls, even no sound for ivr, voicemail greetings, busy message and voice mail messages.

I tried all possibilities and created ticket with digium support as well for g100 they checked and i was told there is no problem with g100. Issue seems to be in asterisk, thing is it's working fine with redfone T1 gateway.

Below is my sip.conf. Do we need to configure any thing specific for incoming audio to work with g100?

[digiumtrunk]
type=friend
username=digiumtrunk
secret=digiumtrunk
host=10.14.0.29(g100 ip)
context=incoming
qualify=yes
disallow=all
allow=gsm
allow=ulaw
nat=yes

I tried with nat yes, no and never options in sip.conf

[Edit by Rusty - Removed inline debug as per the guidelines]

  was:
Hello
We are using asterisk 1.4.10.1 with digium gateway G100

Due to problem with our old redfone T1 gateway we replaced with digium g100 gateway.

After changing the digium G100 Outgoing is working fine but no audio for incoming calls, even no sound for ivr, voicemail greetings, busy message and voice mail messages.

I tried all possibilities and created ticket with digium support as well for g100 they checked and i was told there is no problem with g100. Issue seems to be in asterisk, thing is it's working fine with redfone T1 gateway.

Below is my sip.conf. Do we need to configure any thing specific for incoming audio to work with g100?

[digiumtrunk]
type=friend
username=digiumtrunk
secret=digiumtrunk
host=10.14.0.29(g100 ip)
context=incoming
qualify=yes
disallow=all
allow=gsm
allow=ulaw
nat=yes

I tried with nat yes, no and never options in sip.conf

Below sip debug message
<--- SIP read from 10.14.0.29:5060 --->
INVITE sip:6148564400 at 10.14.0.21 SIP/2.0
Via: SIP/2.0/UDP 10.14.0.29:5060;branch=z9hG4bK397e7edd
Max-Forwards: 70
From: "3125289339" <sip:3125289339 at 10.14.0.21>;tag=as3423061b
To: <sip:6148564400 at 10.14.0.21>
Contact: <sip:3125289339 at 10.14.0.29:5060>
Call-ID: 275173cc43254aeb6a61a981114ddbb3 at 10.14.0.21
CSeq: 102 INVITE
User-Agent: Digium Gateway
Date: Fri, 12 Sep 2014 02:46:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "3125289339" <sip:3125289339 at 10.14.0.21>;party=calling;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 253

v=0
o=root 2146764325 2146764325 IN IP4 10.14.0.29
s=Digium Gateway
c=IN IP4 10.14.0.29
t=0 0
m=audio 10002 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
[Sep 11 10:45:39] VERBOSE[25204] logger.c: --- (15 headers 12 lines) ---
[Sep 11 10:45:39] VERBOSE[25204] logger.c: Sending to 10.14.0.29 : 5060 (no NAT)
[Sep 11 10:45:39] VERBOSE[25204] logger.c: Using INVITE request as basis request - 275173cc43254aeb6a61a981114ddbb3 at 10.14.0.21
[Sep 11 10:45:39] VERBOSE[25204] logger.c: Found peer 'digiumtrunk'
[Sep 11 10:45:39] VERBOSE[25204] logger.c: Found RTP audio format 0
[Sep 11 10:45:39] VERBOSE[25204] logger.c: Found RTP audio format 101
[Sep 11 10:45:39] VERBOSE[25204] logger.c: Peer audio RTP is at port 10.14.0.29:10002
[Sep 11 10:45:39] VERBOSE[25204] logger.c: Found description format PCMU for ID 0
[Sep 11 10:45:39] VERBOSE[25204] logger.c: Found description format telephone-event for ID 101
[Sep 11 10:45:39] VERBOSE[25204] logger.c: Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Sep 11 10:45:39] VERBOSE[25204] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Sep 11 10:45:39] VERBOSE[25204] logger.c: Peer audio RTP is at port 10.14.0.29:10002
[Sep 11 10:45:39] VERBOSE[25204] logger.c: Looking for 6148564400 in incoming (domain 10.14.0.21)
[Sep 11 10:45:39] VERBOSE[25204] logger.c: list_route: hop: <sip:3125289339 at 10.14.0.29:5060>
[Sep 11 10:45:39] VERBOSE[25204] logger.c: 
<--- Transmitting (no NAT) to 10.14.0.29:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.14.0.29:5060;branch=z9hG4bK397e7edd;received=10.14.0.29
From: "3125289339" <sip:3125289339 at 10.14.0.21>;tag=as3423061b
To: <sip:6148564400 at 10.14.0.21>
Call-ID: 275173cc43254aeb6a61a981114ddbb3 at 10.14.0.21
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6148564400 at 10.14.0.21>
Content-Length: 0

<------------>
[Sep 11 10:45:39] VERBOSE[26751] logger.c:     -- Executing [6148564400 at incoming:1] Goto("SIP/digiumtrunk-a35493b0", "ivr|s|1") in new stack
[Sep 11 10:45:39] VERBOSE[26751] logger.c:     -- Goto (ivr,s,1)
[Sep 11 10:45:39] VERBOSE[26751] logger.c:     -- Executing [s at ivr:1] Answer("SIP/digiumtrunk-a35493b0", "") in new stack
[Sep 11 10:45:39] VERBOSE[26751] logger.c: Audio is at 10.14.0.21 port 19982
[Sep 11 10:45:39] VERBOSE[26751] logger.c: Adding codec 0x4 (ulaw) to SDP
[Sep 11 10:45:39] VERBOSE[26751] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Sep 11 10:45:39] VERBOSE[26751] logger.c: 
<--- Reliably Transmitting (no NAT) to 10.14.0.29:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.14.0.29:5060;branch=z9hG4bK397e7edd;received=10.14.0.29
From: "3125289339" <sip:3125289339 at 10.14.0.21>;tag=as3423061b
To: <sip:6148564400 at 10.14.0.21>;tag=as1a141e54
Call-ID: 275173cc43254aeb6a61a981114ddbb3 at 10.14.0.21
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6148564400 at 10.14.0.21>
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 24768 24768 IN IP4 10.14.0.21
s=session
c=IN IP4 10.14.0.21
t=0 0
m=audio 19982 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
[Sep 11 10:45:39] VERBOSE[26751] logger.c:     -- Executing [s at ivr:2] BackGround("SIP/digiumtrunk-a35493b0", "MainGreetings") in new stack
[Sep 11 10:45:39] VERBOSE[25204] logger.c: 
<--- SIP read from 10.14.0.29:5060 --->
ACK sip:6148564400 at 10.14.0.21 SIP/2.0
Via: SIP/2.0/UDP 10.14.0.29:5060;branch=z9hG4bK6289fdcc
Max-Forwards: 70
From: "3125289339" <sip:3125289339 at 10.14.0.21>;tag=as3423061b
To: <sip:6148564400 at 10.14.0.21>;tag=as1a141e54
Contact: <sip:3125289339 at 10.14.0.29:5060>
Call-ID: 275173cc43254aeb6a61a981114ddbb3 at 10.14.0.21
CSeq: 102 ACK
User-Agent: Digium Gateway
Content-Length: 0

> Incoming audio not working
> --------------------------
>
>                 Key: ASTERISK-24320
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24320
>             Project: Asterisk
>          Issue Type: Information Request
>      Security Level: None
>    Affects Versions: 1.8.9.3
>            Reporter: Siva
>
> Hello
> We are using asterisk 1.4.10.1 with digium gateway G100
> Due to problem with our old redfone T1 gateway we replaced with digium g100 gateway.
> After changing the digium G100 Outgoing is working fine but no audio for incoming calls, even no sound for ivr, voicemail greetings, busy message and voice mail messages.
>  
> I tried all possibilities and created ticket with digium support as well for g100 they checked and i was told there is no problem with g100. Issue seems to be in asterisk, thing is it's working fine with redfone T1 gateway.
> Below is my sip.conf. Do we need to configure any thing specific for incoming audio to work with g100?
> [digiumtrunk]
> type=friend
> username=digiumtrunk
> secret=digiumtrunk
> host=10.14.0.29(g100 ip)
> context=incoming
> qualify=yes
> disallow=all
> allow=gsm
> allow=ulaw
> nat=yes
> I tried with nat yes, no and never options in sip.conf
> [Edit by Rusty - Removed inline debug as per the guidelines]

--
This message was sent by Atlassian JIRA
(v6.2#6252)