Re: [asterisk-users] No audio for 10 seconds and then comfort noise

[David Cunningham]

Hi Joshua,

You're right, it was a firewall problem. One of those things where testing
a change in one place throws up a previously unseen problem somewhere else!
Thanks for the tip.


On Thu, 19 May 2022 at 21:18, Joshua C. Colp  wrote:

> On Thu, May 19, 2022 at 6:04 AM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hi Dovid and Joshua,
>>
>> The PSTN is sending RTP immediately after the 200 OK, on both legs of the
>> call. Since the PCAP taken on the Asterisk server itself shows this RTP
>> from the PSTN then presumably it can't be a network issue preventing the
>> RTP.
>>
>> Having said that, the problem is not reproduced when the peer is another
>> Asterisk server on the same network, and that does point to a network
>> difference.
>>
>> Is there any other way in which the RTP keepalive might affect Asterisk's
>> behaviour?
>>
>
> No, the option only does anything if no RTP has been sent for a period of
> time. It doesn't fundamentally alter the behavior of RTP in general.
>
> Another thing to consider is that a PCAP is taken before any local
> firewall rules are applied, which can give a false impression that the
> firewall on the system is not an issue when in reality it can be. That's
> something to check.
>
> --
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
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Re: [asterisk-users] No audio for 10 seconds and then comfort noise

[Joshua C. Colp]

On Thu, May 19, 2022 at 6:04 AM David Cunningham 
wrote:

> Hi Dovid and Joshua,
>
> The PSTN is sending RTP immediately after the 200 OK, on both legs of the
> call. Since the PCAP taken on the Asterisk server itself shows this RTP
> from the PSTN then presumably it can't be a network issue preventing the
> RTP.
>
> Having said that, the problem is not reproduced when the peer is another
> Asterisk server on the same network, and that does point to a network
> difference.
>
> Is there any other way in which the RTP keepalive might affect Asterisk's
> behaviour?
>

No, the option only does anything if no RTP has been sent for a period of
time. It doesn't fundamentally alter the behavior of RTP in general.

Another thing to consider is that a PCAP is taken before any local firewall
rules are applied, which can give a false impression that the firewall on
the system is not an issue when in reality it can be. That's something to
check.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] No audio for 10 seconds and then comfort noise

[David Cunningham]

Hi Dovid and Joshua,

The PSTN is sending RTP immediately after the 200 OK, on both legs of the
call. Since the PCAP taken on the Asterisk server itself shows this RTP
from the PSTN then presumably it can't be a network issue preventing the
RTP.

Having said that, the problem is not reproduced when the peer is another
Asterisk server on the same network, and that does point to a network
difference.

Is there any other way in which the RTP keepalive might affect Asterisk's
behaviour?

Thanks for your help on this.


On Thu, 19 May 2022 at 20:40, Joshua C. Colp  wrote:

> On Thu, May 19, 2022 at 3:52 AM Dovid Bender  wrote:
>
>> David,
>>
>> Are you getting any RTP from the PSTN for either leg? If not it could be
>> that they assume you are behind NAT and want to see where the SRC of the
>> RTP before they send it back. We had a few carriers that did this. The
>> easiest way to get around it was to play a 0.5 second audio clip to the
>> incoming leg. This will send RTP to the inbound carrier, causing them to
>> send RTP back to you which would then hit the terminating carrier, which
>> then sends you back RTP completing the loop. The dialplan looks
>> something like this.
>>
>> same =>n, Progress()
>> same =>n,
>> Playback(/var/lib//asterisk_custom/sounds/xc,noanswer)
>> same =>n, Dial(SIP/+${EXTEN}@carrier,,)
>>
>
> I've also seen this happen due to networking equipment, specifically the
> equipment wanting Asterisk to send packets before allowing packets in. If
> both sides of the call are in this state, then you reach a stalemate and
> media won't flow. Since rtp_keepalive is generated by Asterisk, it gets
> sent, and media starts flowing.
>
> --
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



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Re: [asterisk-users] No audio for 10 seconds and then comfort noise

[Joshua C. Colp]

On Thu, May 19, 2022 at 3:52 AM Dovid Bender  wrote:

> David,
>
> Are you getting any RTP from the PSTN for either leg? If not it could be
> that they assume you are behind NAT and want to see where the SRC of the
> RTP before they send it back. We had a few carriers that did this. The
> easiest way to get around it was to play a 0.5 second audio clip to the
> incoming leg. This will send RTP to the inbound carrier, causing them to
> send RTP back to you which would then hit the terminating carrier, which
> then sends you back RTP completing the loop. The dialplan looks
> something like this.
>
> same =>n, Progress()
> same =>n,
> Playback(/var/lib//asterisk_custom/sounds/xc,noanswer)
> same =>n, Dial(SIP/+${EXTEN}@carrier,,)
>

I've also seen this happen due to networking equipment, specifically the
equipment wanting Asterisk to send packets before allowing packets in. If
both sides of the call are in this state, then you reach a stalemate and
media won't flow. Since rtp_keepalive is generated by Asterisk, it gets
sent, and media starts flowing.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] No audio for 10 seconds and then comfort noise

[Dovid Bender]

David,

Are you getting any RTP from the PSTN for either leg? If not it could be
that they assume you are behind NAT and want to see where the SRC of the
RTP before they send it back. We had a few carriers that did this. The
easiest way to get around it was to play a 0.5 second audio clip to the
incoming leg. This will send RTP to the inbound carrier, causing them to
send RTP back to you which would then hit the terminating carrier, which
then sends you back RTP completing the loop. The dialplan looks
something like this.

same =>n, Progress()
same =>n,
Playback(/var/lib//asterisk_custom/sounds/xc,noanswer)
same =>n, Dial(SIP/+${EXTEN}@carrier,,)






On Thu, May 19, 2022 at 12:13 AM David Cunningham 
wrote:

> We found that the 10 seconds relates to the "rtpkeepalive =10" in our
> sip.conf. If the rtpkeepalive is reduced then the delay reduces as well. If
> rtpkeepalive is removed from sip.conf then audio never starts flowing.
>
> Does that help anyone make sense of what's happening?
>
> We have DAHDI running on the server:
>
> # asterisk -rx 'dahdi show version'
> DAHDI Version: 3.0.0 Echo Canceller:
> # asterisk -rx 'dahdi show status'
> Description  Alarms  IRQbpviol CRCFra
> Codi Options  LBO
>
>
> On Thu, 19 May 2022 at 15:51, David Cunningham 
> wrote:
>
>> Hello,
>>
>> We are running an Asterisk 13 server which is having a strange problem,
>> where on calls which are received from the PSTN and then forwarded out to
>> the PSTN again there is no audio for the first 10 seconds of the call. At
>> the 10 second mark audio starts flowing fine, and in a PCAP we see that it
>> starts with a few "comfort noise" packers before the real audio starts.
>>
>> It can be reproduced with a very simple extension like this:
>> exten => 4081234567, 2, Dial(SIP/6501234...@bb.bb.bb.138)
>> where 4081234567 is the number we receive the call on, and 6501234567 is
>> the number we're forwarding it out to.
>>
>> In the Asterisk log we don't see any obvious reason for the audio to
>> start flowing at the 10 second mark. All that is logged at that time is the
>> following below.
>>
>> Would anyone have any ideas? Historically Asterisk didn't generate
>> comfort noise - has that changed in version 13?
>>
>> [May 17 20:26:24] VERBOSE[11933] res_rtp_asterisk.c: Sent Comfort Noise
>> RTP packet to aa.aa.aa.76:64280 (type 02, seq 009268, ts 00, len 01)
>> [May 17 20:26:24] VERBOSE[17794][C-0027] res_rtp_asterisk.c: Got RTP
>> packet from aa.aa.aa.76:64280 (type 00, seq 000662, ts 105920, len 000160)
>> [May 17 20:26:24] DEBUG[17725][C-0027] res_rtp_asterisk.c: Ooh,
>> format changed from none to ulaw
>> [May 17 20:26:24] DEBUG[17725][C-0027] res_rtp_asterisk.c: Starting
>> RTCP transmission on RTP instance '0x14f4cc025998'
>> [May 17 20:26:24] VERBOSE[17725][C-0027] res_rtp_asterisk.c: Sent RTP
>> packet to bb.bb.bb.20:35412 (type 00, seq 020934, ts 105920, len 000160)
>> [May 17 20:26:24] VERBOSE[17725][C-0027] res_rtp_asterisk.c: Got RTP
>> packet from bb.bb.bb.20:35412 (type 00, seq 029996, ts 102760, len 000160)
>>
>> Thanks very much,
>>
>> --
>> David Cunningham, Voisonics Limited
>> http://voisonics.com/
>> USA: +1 213 221 1092
>> New Zealand: +64 (0)28 2558 3782
>>
>
>
> --
> David Cunningham, Voisonics Limited
> http://voisonics.com/
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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Re: [asterisk-users] No audio for 10 seconds and then comfort noise

[David Cunningham]

We found that the 10 seconds relates to the "rtpkeepalive =10" in our
sip.conf. If the rtpkeepalive is reduced then the delay reduces as well. If
rtpkeepalive is removed from sip.conf then audio never starts flowing.

Does that help anyone make sense of what's happening?

We have DAHDI running on the server:

# asterisk -rx 'dahdi show version'
DAHDI Version: 3.0.0 Echo Canceller:
# asterisk -rx 'dahdi show status'
Description  Alarms  IRQbpviol CRCFra
Codi Options  LBO


On Thu, 19 May 2022 at 15:51, David Cunningham 
wrote:

> Hello,
>
> We are running an Asterisk 13 server which is having a strange problem,
> where on calls which are received from the PSTN and then forwarded out to
> the PSTN again there is no audio for the first 10 seconds of the call. At
> the 10 second mark audio starts flowing fine, and in a PCAP we see that it
> starts with a few "comfort noise" packers before the real audio starts.
>
> It can be reproduced with a very simple extension like this:
> exten => 4081234567, 2, Dial(SIP/6501234...@bb.bb.bb.138)
> where 4081234567 is the number we receive the call on, and 6501234567 is
> the number we're forwarding it out to.
>
> In the Asterisk log we don't see any obvious reason for the audio to start
> flowing at the 10 second mark. All that is logged at that time is the
> following below.
>
> Would anyone have any ideas? Historically Asterisk didn't generate comfort
> noise - has that changed in version 13?
>
> [May 17 20:26:24] VERBOSE[11933] res_rtp_asterisk.c: Sent Comfort Noise
> RTP packet to aa.aa.aa.76:64280 (type 02, seq 009268, ts 00, len 01)
> [May 17 20:26:24] VERBOSE[17794][C-0027] res_rtp_asterisk.c: Got RTP
> packet from aa.aa.aa.76:64280 (type 00, seq 000662, ts 105920, len 000160)
> [May 17 20:26:24] DEBUG[17725][C-0027] res_rtp_asterisk.c: Ooh, format
> changed from none to ulaw
> [May 17 20:26:24] DEBUG[17725][C-0027] res_rtp_asterisk.c: Starting
> RTCP transmission on RTP instance '0x14f4cc025998'
> [May 17 20:26:24] VERBOSE[17725][C-0027] res_rtp_asterisk.c: Sent RTP
> packet to bb.bb.bb.20:35412 (type 00, seq 020934, ts 105920, len 000160)
> [May 17 20:26:24] VERBOSE[17725][C-0027] res_rtp_asterisk.c: Got RTP
> packet from bb.bb.bb.20:35412 (type 00, seq 029996, ts 102760, len 000160)
>
> Thanks very much,
>
> --
> David Cunningham, Voisonics Limited
> http://voisonics.com/
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782
>


-- 
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http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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[asterisk-users] No audio for 10 seconds and then comfort noise

[David Cunningham]

Hello,

We are running an Asterisk 13 server which is having a strange problem,
where on calls which are received from the PSTN and then forwarded out to
the PSTN again there is no audio for the first 10 seconds of the call. At
the 10 second mark audio starts flowing fine, and in a PCAP we see that it
starts with a few "comfort noise" packers before the real audio starts.

It can be reproduced with a very simple extension like this:
exten => 4081234567, 2, Dial(SIP/6501234...@bb.bb.bb.138)
where 4081234567 is the number we receive the call on, and 6501234567 is
the number we're forwarding it out to.

In the Asterisk log we don't see any obvious reason for the audio to start
flowing at the 10 second mark. All that is logged at that time is the
following below.

Would anyone have any ideas? Historically Asterisk didn't generate comfort
noise - has that changed in version 13?

[May 17 20:26:24] VERBOSE[11933] res_rtp_asterisk.c: Sent Comfort Noise RTP
packet to aa.aa.aa.76:64280 (type 02, seq 009268, ts 00, len 01)
[May 17 20:26:24] VERBOSE[17794][C-0027] res_rtp_asterisk.c: Got RTP
packet from aa.aa.aa.76:64280 (type 00, seq 000662, ts 105920, len 000160)
[May 17 20:26:24] DEBUG[17725][C-0027] res_rtp_asterisk.c: Ooh, format
changed from none to ulaw
[May 17 20:26:24] DEBUG[17725][C-0027] res_rtp_asterisk.c: Starting
RTCP transmission on RTP instance '0x14f4cc025998'
[May 17 20:26:24] VERBOSE[17725][C-0027] res_rtp_asterisk.c: Sent RTP
packet to bb.bb.bb.20:35412 (type 00, seq 020934, ts 105920, len 000160)
[May 17 20:26:24] VERBOSE[17725][C-0027] res_rtp_asterisk.c: Got RTP
packet from bb.bb.bb.20:35412 (type 00, seq 029996, ts 102760, len 000160)

Thanks very much,

-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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Re: [asterisk-users] no audio both ways with ipv6

[hw]

On Thu, 2021-10-14 at 21:21 +0200, Antony Stone wrote:
> On Thursday 14 October 2021 at 19:22:00, hw wrote:
> 
> > Hi,
> > 
> > when asterisk registers with the VOIP provider via ipv6 and when
> > local phones don't work with ipv6 but only with ipv4, am I to
> > expect issues?
> 
> Do a SIP packet capture and see what the SDP in the INVITE is telling each 
> end 
> to expect from the other.

Hmm I could try that maybe, as a last resort.

> > I'm receiving incoming calls via the provider, asterisk correctly
> > dials the phone where the calls are suposed to go to, the phone
> > rings --- and when I pick it up, there is no audio in either direction.
> 
> Sounds like the setup is trying to do direct media - which obviously cannot 
> work between an IPv4-only phone and an IPv6-only provider.
> 
> Make sure Asterisk remains in the audio path and it should "almost transcode" 
> for you.

I thought about that, and I think direct media isn't being used.  It works with
ipv4, and if it was using direct media, ipv4 wouldn't work, either.  IIRC I
tried with 'aor (or endpoint?)/direct_media = no'.  Unfortunately, I can't 
really
make test calls to try things out.

When a call comes in and I pick up the phone, asterisk says it has learned the 
ipv6
address of the VOIP provider on one side and the ipv4 address on the other, and 
the
channels are joining a simple bridge --- whatever that means.  Is there 
something
that would tell me if asterisk is trying to set up direct media or remains in
between?

> I have audio working over just such an arrangement (in my case, an IPv4-only 
> provider, and phones connected via IPv6) without problems.

I wish I could try with an ipv6 phone, but I couldn't get my Polycom VVX 1500D 
to
work with ipv6 at all.  It was suggested on the Polycom forum that the firmware
is too old and that I update to the latest, but the latest doesn't run on this 
phone
because the phone is too old.  The release before the latest is supposed to 
work,
but that is nowhere to be found, and I didn't get any more answers.

I tried Twinkle on my computer, but that doesn't support ipv6 at all.  There 
must
be something special going on with ipv6 when it comes to SIP and/or RTP.


> 
> Antony.
> 
> -- 
> The difference between theory and practice is that in theory there is no 
> difference, whereas in practice there is.
> 
>Please reply to the list;
>  please *don't* CC me.
> 



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Re: [asterisk-users] no audio both ways with ipv6

[Antony Stone]

On Thursday 14 October 2021 at 19:22:00, hw wrote:

> Hi,
> 
> when asterisk registers with the VOIP provider via ipv6 and when
> local phones don't work with ipv6 but only with ipv4, am I to
> expect issues?

Do a SIP packet capture and see what the SDP in the INVITE is telling each end 
to expect from the other.

> I'm receiving incoming calls via the provider, asterisk correctly
> dials the phone where the calls are suposed to go to, the phone
> rings --- and when I pick it up, there is no audio in either direction.

Sounds like the setup is trying to do direct media - which obviously cannot 
work between an IPv4-only phone and an IPv6-only provider.

Make sure Asterisk remains in the audio path and it should "almost transcode" 
for you.

I have audio working over just such an arrangement (in my case, an IPv4-only 
provider, and phones connected via IPv6) without problems.


Antony.

-- 
The difference between theory and practice is that in theory there is no 
difference, whereas in practice there is.

   Please reply to the list;
 please *don't* CC me.

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[asterisk-users] no audio both ways with ipv6

[hw]

Hi,

when asterisk registers with the VOIP provider via ipv6 and when
local phones don't work with ipv6 but only with ipv4, am I to
expect issues?

I'm receiving incoming calls via the provider, asterisk correctly
dials the phone where the calls are suposed to go to, the phone
rings --- and when I pick it up, there is no audio in either direction.

There are no packets showing up in the logs as being rejected by the
firewall.  I can make outgoing calls just fine, and those are with
the VOIP provider on one side and the same ipv4 phone on the other.

How can it be that incoming calls have no audio?




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[asterisk-users] No audio after hold

[arish haque]

Hi all,

No audio flows after hold

I'm getting this below error after unholding the call.

chan_sip.c:10088 process_sdp: Declining non-primary audio stream:
audio 14670 UDP/TLS/RTP/SAVPF 107 103 104 9 0 8 106 105 13 110 112 113 101

is it a codec issue or something else?


With Regards,

Arish
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[asterisk-users] No audio on direct call from trunk to SPA-8000

[Carlos Chavez]

    I am having one of those days.  We just replaced an old Asterisk 
1.8 server with a new Asterisk 13.21.1 (both using Freepbx) and almost 
everything is working except for some incoming calls directed to a Cisco 
SPA-8000.  The PSTN trunk is SIP.  Only calls coming from the PSTN to a 
direct DID that just rings an extension on the SPA get no incoming 
audio.  All other calls, including calls from the PSTN that go through 
the main IVR or operator have audio.


    I made sure that the trunk has direct_media=no.  I checked the SPA 
configuration to make sure it is not using NAT.  Only the SPA suffers 
from this as regular SIP phones can receive calls from their DID with no 
problems.  This is the first time I use an SPA analog adapter with 
PJSIP.  They work great with chan_sip so I do not know what maybe wrong 
here.  Anyone using an SPA-8000 with PJSIP?  Any settings I should check 
on the SPA or in Asterisk?


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Carlos Chávez
+52 (55)8116-9161


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Re: [asterisk-users] No audio when using TLS/SRTP with Kamailio and Asterisk 13

[Matthew Jordan]

On Tue, Aug 18, 2015 at 3:12 AM, Chirag Desai  wrote:

> Hi all,
>
> I'm using Kamailio + Asterisk 13 (PJSIP), where Kamailio (using rtpengine)
> acts as the registrar and forwards all calls to Asterisk.
>
> This works fine when using udp / tcp and RTP. When switching to TLS/SRTP,
> the call is set up correctly, however, I get no audio.
>
> When I skip kamailio and connect my two endpoints to asterisk directly I
> get a perfect call with SRTP.
>
> The same is also true when I skip asterisk and have the call handled by
> Kamailio (using RTPEngine).
>
> In PJSIP my transports look like this:
>
> [transport-tcp]
> type=transport
> protocol=tcp;udp,tcp,tls,ws,wss
> bind=0.0.0.0:5060
> local_net=[asterisk local ip]/17
> external_media_address=[asterisk external ip]
> external_signaling_address=[asterisk external ip]
>
> [transport-tls]
> type=transport
> protocol=tls
> bind=0.0.0.0:5063
> ca_list_file=/etc/asterisk/certificates/cert.crt
> cert_file=/etc/asterisk/certificates/certificate.crt
> priv_key_file=/etc/asterisk/certificates/key.key
> method=tlsv1
>
>
> My endpoint looks like this:
>
> [kamailio]
> type=endpoint
> context=kam_out
> disallow=all
> allow=alaw
> allow=g722
> allow=ulaw
> allow=gsm
> aors=kamailio
> direct_media=no
> media_encryption=sdes
> media_address=[Asterisk Local IP]
> rtp_symmetric=yes
> force_rport=no
> rewrite_contact=yes
> outbound_proxy=sip:[Kamailio Local IP]:5060\;transport=tcp\;lr
>
> [kamailio]
> type=identify
> endpoint=kamailio
> match=[Kamailio Local IP]/17
>
> [kamailio]
> type=aor
> contact=sip:[Kamailio Local IP]:5060\;transport=tcp
>
>
> My dialplan looks like this
>
> [kam_out]
>
> exten => 1001,1,Playback(demo-echotest)  ; Let them know what's going on
> same => n,Echo ; Do the echo test
> same => n,Playback(demo-echodone)  ; Let them know it's over
> same => n,Hangup()
>
>
> exten => _kb-.,1,NoOp(Calling a registred user with number ${EXTEN})
> same => n,Set(callee=${PJSIP_HEADER(read,To)})
> same => n,Set(callee=${callee:5})
> same => n,Set(callee=${callee:0:-1}) ; removes the >
> same => n,Dial(PJSIP/kamailio/sip:${callee})
> same => n,Hangup()
>
> When a call comes via kamailio it comes with a prefix of 'kb' if the value
> is an extension e.g. 1000 - 1999. Otherwise users can dial a prefix of 45
> e.g. 451001 to hit the Echo Test.
>
> As mentioned the echo test works fine, however the actual call between two
> endpoints has no audio. RTP debug shows nothing. PJSIP shows two channels
> in a simple bridge, but no sound. Usually PJSIP says RTP Probation passed
> and shows the IP address but in this case it does not.
>
>
The PJSIP stack only provides SIP signalling; it doesn't interfere with the
media handling in Asterisk. The handling of media is done by the RTP engine
implementation, res_rtp_asterisk.

I don't think this is a problem, however, with res_rtp_asterisk or
Asterisk. If RTP debug doesn't show any traffic, then Asterisk is almost
certainly not receiving any media.

What does a PCAP show? I'd look at where the RTPEngine is forwarding your
RTP packets off to, and see if they are getting sent somewhere other than
Asterisk.



> I'm guessing the issue is something funny in PJSIP, although I'm not 100%
> since it does work when I turn SRTP and TLS off.
>
> For testing I'm using CsipSimple and a Snom 760. Both are set with SRTP
> mandatory and are using TLS to talk to Kamailio.
>
> When kamailio talks to asterisk it uses TCP over a local network.
>
> I've been pulling my hair out for days. I really would appreciate any
> ideas or some pointing in the right direction here.
>
> Thanks in advance,
>
> C
>
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>



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[asterisk-users] No audio when using TLS/SRTP with Kamailio and Asterisk 13

[Chirag Desai]

Hi all,

I'm using Kamailio + Asterisk 13 (PJSIP), where Kamailio (using rtpengine)
acts as the registrar and forwards all calls to Asterisk.

This works fine when using udp / tcp and RTP. When switching to TLS/SRTP,
the call is set up correctly, however, I get no audio.

When I skip kamailio and connect my two endpoints to asterisk directly I
get a perfect call with SRTP.

The same is also true when I skip asterisk and have the call handled by
Kamailio (using RTPEngine).

In PJSIP my transports look like this:

[transport-tcp]
type=transport
protocol=tcp;udp,tcp,tls,ws,wss
bind=0.0.0.0:5060
local_net=[asterisk local ip]/17
external_media_address=[asterisk external ip]
external_signaling_address=[asterisk external ip]

[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0:5063
ca_list_file=/etc/asterisk/certificates/cert.crt
cert_file=/etc/asterisk/certificates/certificate.crt
priv_key_file=/etc/asterisk/certificates/key.key
method=tlsv1


My endpoint looks like this:

[kamailio]
type=endpoint
context=kam_out
disallow=all
allow=alaw
allow=g722
allow=ulaw
allow=gsm
aors=kamailio
direct_media=no
media_encryption=sdes
media_address=[Asterisk Local IP]
rtp_symmetric=yes
force_rport=no
rewrite_contact=yes
outbound_proxy=sip:[Kamailio Local IP]:5060\;transport=tcp\;lr

[kamailio]
type=identify
endpoint=kamailio
match=[Kamailio Local IP]/17

[kamailio]
type=aor
contact=sip:[Kamailio Local IP]:5060\;transport=tcp


My dialplan looks like this

[kam_out]

exten => 1001,1,Playback(demo-echotest)  ; Let them know what's going on
same => n,Echo ; Do the echo test
same => n,Playback(demo-echodone)  ; Let them know it's over
same => n,Hangup()


exten => _kb-.,1,NoOp(Calling a registred user with number ${EXTEN})
same => n,Set(callee=${PJSIP_HEADER(read,To)})
same => n,Set(callee=${callee:5})
same => n,Set(callee=${callee:0:-1}) ; removes the >
same => n,Dial(PJSIP/kamailio/sip:${callee})
same => n,Hangup()

When a call comes via kamailio it comes with a prefix of 'kb' if the value
is an extension e.g. 1000 - 1999. Otherwise users can dial a prefix of 45
e.g. 451001 to hit the Echo Test.

As mentioned the echo test works fine, however the actual call between two
endpoints has no audio. RTP debug shows nothing. PJSIP shows two channels
in a simple bridge, but no sound. Usually PJSIP says RTP Probation passed
and shows the IP address but in this case it does not.

I'm guessing the issue is something funny in PJSIP, although I'm not 100%
since it does work when I turn SRTP and TLS off.

For testing I'm using CsipSimple and a Snom 760. Both are set with SRTP
mandatory and are using TLS to talk to Kamailio.

When kamailio talks to asterisk it uses TCP over a local network.

I've been pulling my hair out for days. I really would appreciate any ideas
or some pointing in the right direction here.

Thanks in advance,

C
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[asterisk-users] No audio on SIP over WebRTC

[Vinicius Fontes]

I'm following this tutorial (
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5) to
deploy WebRTC support but I'm having an issue with RTP when the WebRTC
softphone is behind NAT.

In my scenario, the Asterisk server is running a public IPv4, and the
softphone is behind NAT. I can register and make a call normally, but I
don't get any audio in neither way (Asterisk/softphone and
softphone/Asterisk). Using the very same config files but having the
softphone and Asterisk on the same network it works fine.

Any tips on how to solve this? Here's my relevant files.

*;sip.conf:*
[general]
udpbindaddr=0.0.0.0:5060
realm=10.201.0.106 ;replace with your Asterisk server public IP address or
host
transport=udp,ws,wss
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1

[6000]
host=dynamic
secret=mysecret
context=default
type=friend
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
qualify=yes

[6001]
host=dynamic
secret=mysecret
context=default
type=friend
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=no
disallow=all
allow=ulaw
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass


*extensions.conf:*
[default]
exten => _6XXX,1,Dial(SIP/${EXTEN})


*rtp.conf:*
[general]
rtpstart=1
rtpend=2
icesupport=yes
stunaddr=stun.l.google.com:19302
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Re: [asterisk-users] no audio from meetme conference bridge

[Vik Killa]

I may have discovered what the issue was... we had *rfc2833compensate=yes*
on version 1.8 of asterisk, I believe the extra RTP asterisk was sending
with this enabled caused a problem. Can anyone confirm this? I'm not
exactly sure (in technical terms) what this setting does...I only found a
description that it's necessary for asterisk version<=1.4

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[asterisk-users] no audio from meetme conference bridge

[Vik Killa]

Asterisk intermittently does not send audio back to the callers in the
meetme conference bridge. If the caller hangs up and calls back sometimes
the audio will work and sometimes it does not. We have taken packet
captures and reviewed the SIP and SDP, both are correct and you can
actually hear the audio being transmitted from the callers to the
conference bridge but no audio is sent back to the callers. We've also
verified the meetme conference flags are correct so the callers are not set
to be muted. I should mention this only happens when the call is going from
one asterisk box to another (the second box being where the conference is
living). If all conference participants are on single asterisk machine the
problem does not occur. Has anybody had this problem in past? I could not
find the issue doing a google.
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Re: [asterisk-users] No audio until you put call on hold...

[Matthew J. Roth]

Carlos Chavez wrote:
>
> I have been struggling with an audio issue for a week now and have
> not been able to solve it.
>
> We have an Asterisk server (now running 11.4 but started with 1.8)
> with several sip phones on an internal network and a SIP trunk for
> external calls.  We recently put several phones in service that
> connect via the Internet to the server.  All NAT settings and port
> configurations were done and all phones register.  The problem we have
> is that when external phones dial a pstn number they get no audio.  We
> found that if you dial and put the call on hold for a couple second
> you then get audio on the call.
>
> I really do not know what else I can check in the configuration.  Why
> would putting the call on hold get the audio flowing?  Any ideas or
> recommendations?


Carlos,

Please provide SIP traces of both call legs (external phone to Asterisk and
Asterisk to SIP trunk) annotated to show when the audio starts as well as the
CLI output of 'sip show settings', 'sip show peer ', and 'sip
show peer '.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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[asterisk-users] No audio until you put call on hold...

[Carlos Chavez]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

I have been struggling with an audio issue for a week now and have
not been able to solve it.

We have an Asterisk server (now running 11.4 but started with 1.8)
with several sip phones on an internal network and a SIP trunk for
external calls.  We recently put several phones in service that
connect via the Internet to the server.  All NAT settings and port
configurations were done and all phones register.  The problem we have
is that when external phones dial a pstn number they get no audio.  We
found that if you dial and put the call on hold for a couple second
you then get audio on the call.

I really do not know what else I can check in the configuration.  Why
would putting the call on hold get the audio flowing?  Any ideas or
recommendations?

- -- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
-BEGIN PGP SIGNATURE-
Version: GnuPG/MacGPG2 v2.0.18 (Darwin)
Comment: GPGTools - http://gpgtools.org
Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/

iEYEARECAAYFAlG5+sUACgkQqmNh+MyHzx56+wCfWCLoqlm3Loviat2zJJWbKsL+
Om4AoKI+/db48174uetU+2DAjvcP1S2c
=qmvr
-END PGP SIGNATURE-

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Re: [asterisk-users] no audio while call forwarding, yes audio with followme

[Bart Coninckx]

Hi.

Thank you.

You mean do each call separately? That works without a glitch, nothing 
peculiar.


Thx,


BC



On 09/25/12 23:28, Danny Nicholas wrote:


Do the call both ways again and check(post) the CLI output.

*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bart 
Coninckx

*Sent:* Tuesday, September 25, 2012 4:23 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] no audio while call forwarding, yes audio 
with followme


Hi all,

the subject says it all.
Technical details:
- Asterisk 1.8.7.1
- Behind NAT
- Using external SIP provider

The call forwarding is tested both with this functionality on the 
phone and with configuration in the dialplan. In the latter case a 
database variable is set to the external number, if set a Dial command 
calls this number. So really nothing fancy (actually I followed the 
example on http://www.voip-info.org/wiki/view/Asterisk+call+forwarding ).


sip.conf has nat=yes, externip= ... and I tried every setting of 
directmedia in the providers configuration part.


Followme works flawlessly, so I'm really wondering if this is a NAT 
issue.



Can anyone point me into a certain direction?


Thx


BC



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Re: [asterisk-users] no audio while call forwarding, yes audio with followme

[Danny Nicholas]

Do the call both ways again and check(post) the CLI output.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bart Coninckx
Sent: Tuesday, September 25, 2012 4:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] no audio while call forwarding, yes audio with
followme

 

Hi all,

the subject says it all.
Technical details: 
- Asterisk 1.8.7.1 
- Behind NAT 
- Using external SIP provider

The call forwarding is tested both with this functionality on the phone and
with configuration in the dialplan. In the latter case a database variable
is set to the external number, if set a Dial command calls this number. So
really nothing fancy (actually I followed the example on
http://www.voip-info.org/wiki/view/Asterisk+call+forwarding ).

sip.conf has nat=yes, externip= ... and I tried every setting of directmedia
in the providers configuration part. 

Followme works flawlessly, so I'm really wondering if this is a NAT issue. 


Can anyone point me into a certain direction?


Thx


BC

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[asterisk-users] no audio while call forwarding, yes audio with followme

[Bart Coninckx]

Hi all,

the subject says it all.
Technical details:
- Asterisk 1.8.7.1
- Behind NAT
- Using external SIP provider

The call forwarding is tested both with this functionality on the phone 
and with configuration in the dialplan. In the latter case a database 
variable is set to the external number, if set a Dial command calls this 
number. So really nothing fancy (actually I followed the example on 
http://www.voip-info.org/wiki/view/Asterisk+call+forwarding ).


sip.conf has nat=yes, externip= ... and I tried every setting of 
directmedia in the providers configuration part.


Followme works flawlessly, so I'm really wondering if this is a NAT issue.


Can anyone point me into a certain direction?


Thx


BC

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Re: [asterisk-users] No audio playing back voicemail from odbc

[Support]

Well, it gets even stranger

I've installed version 10.2.1, instead of 10.7.1, and copied the configuration 
files from another identical server that is running 10.2.1 is working 
correctly.

I STILL can't get voicemail to play back.  I can hear the password prompts

Theses are, what I think to be, the relevant settings in voicemail.conf:

;minmessage=3
maxsilence=10
silencethreshold=128

When I set silencethresholdo either 500 or 64, I still didn't hear anything.  
(But I did hear several seconds of actual silence.  The .wav file contains 
nothing but silence.

So, fiddling with the silencethreshold in both directions, doesn't seem to 
change the symptoms.

Where else should I look?

TIA

Mike.

On Saturday 28 July 2012 6:45:15 pm Matthew Jordan wrote:
> - Original Message -
> 
> > From: "Support" 
> > To: asterisk-users@lists.digium.com
> > Cc: "Matthew Jordan" 
> > Sent: Saturday, July 28, 2012 2:38:09 PM
> > Subject: Re: [asterisk-users] No audio playing back voicemail from odbc
> > 
> > 
> > CLI> core show translation paths slin
> > --- Translation paths SRC Codec "slin" sample rate 8000 ---
> > 
> > slin   To g723  : No Translation Path
> > slin   To gsm   : (slin)->(gsm)
> > slin   To ulaw  : (slin)->(ulaw)
> > 
> > I'm using the ulaw audio codec and wav for storage, so this SHOULD
> > work
> 
> If you're getting a duration of 0, I have to wonder if your
> silencethreshold is playing a factor here.  Asterisk may be treating the
> entire recording as silence.  What happens if you set the maxsilence to
> some valid integer value?  Does it end the voicemail messages while you're
> "leaving" it?  If so, that might indicate that it isn't detecting any
> sound.
> 
> If you set minduration/maxsilence and Asterisk starts killing recordings
> and not saving files, that will also tell you if Asterisk believes the
> recordings are mostly silence.
> 
> > I don't, but this configuration worked before I upgraded from 1.6.x
> > to 10.x.  I
> > should have mentioned that this was part of an upgrade, but it was
> > late, and I
> > was tired.
> > 
> > So, is there something I'm missing?
> 
> I'm not sure.  I'd be curious to see your voicemail.conf.
> 
> > > File storage is the only mechanism to have video voicemail (with
> > > audio)
> > > at this time.
> > 
> > Is there any interest in fixing this situation?  It doesn't seem like
> > it would
> > be too difficult.  I wouldn't mind helping if there is an effort
> > already
> > underway.
> 
> There has been some interest expressed from users, but no development
> plans have been put into place for this feature.
> 
> There are a couple of reasons for that: while it would be possible
> to have multiple formats stored in ODBC/IMAP backends, that doesn't solve
> all of the problems with associating an audio file with a video file.
> For example, some soft phones allow you to start the video media stream
> after the audio media stream has already begun.  This will work fine
> during the video call; however, if the video/audio is stored as a voicemail
> message, Asterisk has no way to associate the beginning of the video file
> with some arbitrary point in the audio file.  Hence, when the video
> message is played back, the video will be out of sync with the audio -
> both are played back starting at the same time, but the soft phone didn't
> start sending the video at the beginning of the audio.
> 
> The solution to this would be to store the audio/video as a single file
> in a media container (such as Matroska).  Not only does this solve the
> audio/video sync issue, but now you don't need to store more then a single
> file in a storage backend.  Unfortunately, this is an extremely non-trivial
> effort.
> 
> --
> Matthew Jordan
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
> 
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-- 

Mike Diehl.

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Re: [asterisk-users] No audio playing back voicemail from odbc

[Support]

Well, it gets even stranger

I've installed version 10.2.1, instead of 10.7.1, and copied the configuration 
files from another identical server that is running 10.2.1 is working 
correctly.

I STILL can't get voicemail to play back.  I can hear the password prompts

Theses are, what I think to be, the relevant settings in voicemail.conf:

;minmessage=3
maxsilence=10
silencethreshold=128

When I set silencethresholdo either 500 or 64, I still didn't hear anything.  
(But I did hear several seconds of actual silence.  The .wav file contains 
nothing but silence.

So, fiddling with the silencethreshold in both directions, doesn't seem to 
change the symptoms.

Where else should I look?

TIA

Mike.


On Saturday 28 July 2012 3:43:55 am Support wrote:
> Hi all,
> 
> I'm trying to get my voicemail messages stored in a mysql database via
> odbc. Most of it is working, except, when I play my
> voicemail messages, I don't hear anything.
> 
> I can confirm that the messages are getting stored in the database:
> 
> select
> uniqueid,msgnum,dir,context,macrocontext,callerid,origtime,duration,mailbox
> user,mailboxcontext,flag,`read` from voicemessages;
> +--++-+
> ---+--+++--+---
> --++--+--+
> 
> | uniqueid | msgnum | dir |
> 
> context   | macrocontext | callerid   | origtime   | duration |
> mailboxuser | mailboxcontext | flag | read |
> +--++-+
> ---+--+++--+---
> --++--+--+
> 
> |7 |  0 | /var/spool/asterisk/voicemail/diehlnet/7001/Old |
> 
> customers |  | 17442025-1 | 1343462922 | 0| 7001
> 
> | diehlnet   |  |0 |
> 
> +--++-+
> ---+--+++--+---
> --++--+--+
> 
> select length(recording) from voicemessages;
> +---+
> 
> | length(recording) |
> 
> +---+
> 
> | 52204 |
> 
> +---+
> 
> This is what the console displays during message playback:
> 
> [Jul 28 02:28:35]   == Parsing
> '/var/spool/asterisk/voicemail/diehlnet/7001/Old/msg.txt': [Jul 28
> 02:28:35]   == Found
> [Jul 28 02:28:35] --  Playing 'vm-
> message.ulaw' (language 'en')
> [Jul 28 02:28:36] --  Playing 'vm-unknown-
> caller.ulaw' (language 'en')
> [Jul 28 02:28:38] --  Playing
> '/var/spool/asterisk/voicemail/diehlnet/7001/Old/msg.slin' (language
> 'en')
> 
> Of course mg.txt and msg.slin don't exist.  I'm assuming Asterisk
> creates them and deletes them?
> 
> But I don't hear anything.  Any ideas?  Asterisk version 10.6.1
> 
> Also, on a side note, ODBC storage and video voicemail aren't going to work
> together, right?

-- 

Take care and have fun,
Mike Diehl.

-- 

Mike Diehl.

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Re: [asterisk-users] No audio playing back voicemail from odbc

[Matthew Jordan]

- Original Message -
> From: "Mike Diehl" 
> To: asterisk-users@lists.digium.com
> Cc: "Matthew Jordan" 
> Sent: Sunday, July 29, 2012 3:39:05 PM
> Subject: Re: [asterisk-users] No audio playing back voicemail from odbc
> 
> On Saturday 28 July 2012 6:45:15 pm Matthew Jordan wrote:
> 
> An alternative might be to store time offsets from start of call in
> the
> database.  Then the playback routines would simply have to play the
> media "in
> order."  Do you see a flaw in this design?

Not really a flaw, but there'd be a lot more to it then that.  You'd
have to modify Playback (or some other application) to interleave the
frames properly from two files that have different formats.  That kind
of mechanism doesn't currently exist.

Not impossible, but certainly more then just knowing which media file
started when.

The nice thing about a media container is that it should have an
API that helps you with those kinds of things.

--
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] No audio playing back voicemail from odbc

[Matthew Jordan]

- Original Message -
> From: "Support" 
> To: asterisk-users@lists.digium.com
> Cc: "Matthew Jordan" 
> Sent: Saturday, July 28, 2012 2:38:09 PM
> Subject: Re: [asterisk-users] No audio playing back voicemail from odbc
> 

> CLI> core show translation paths slin
> --- Translation paths SRC Codec "slin" sample rate 8000 ---
> slin   To g723  : No Translation Path
> slin   To gsm   : (slin)->(gsm)
> slin   To ulaw  : (slin)->(ulaw)
> 
> I'm using the ulaw audio codec and wav for storage, so this SHOULD
> work
> 

If you're getting a duration of 0, I have to wonder if your silencethreshold
is playing a factor here.  Asterisk may be treating the entire recording
as silence.  What happens if you set the maxsilence to some valid integer
value?  Does it end the voicemail messages while you're "leaving" it?  If so,
that might indicate that it isn't detecting any sound.

If you set minduration/maxsilence and Asterisk starts killing recordings
and not saving files, that will also tell you if Asterisk believes the
recordings are mostly silence.

> 
> I don't, but this configuration worked before I upgraded from 1.6.x
> to 10.x.  I
> should have mentioned that this was part of an upgrade, but it was
> late, and I
> was tired.
> 
> So, is there something I'm missing?

I'm not sure.  I'd be curious to see your voicemail.conf.

> > File storage is the only mechanism to have video voicemail (with
> > audio)
> > at this time.
> 
> Is there any interest in fixing this situation?  It doesn't seem like
> it would
> be too difficult.  I wouldn't mind helping if there is an effort
> already
> underway.
> 

There has been some interest expressed from users, but no development
plans have been put into place for this feature.

There are a couple of reasons for that: while it would be possible
to have multiple formats stored in ODBC/IMAP backends, that doesn't solve
all of the problems with associating an audio file with a video file.
For example, some soft phones allow you to start the video media stream
after the audio media stream has already begun.  This will work fine
during the video call; however, if the video/audio is stored as a voicemail
message, Asterisk has no way to associate the beginning of the video file
with some arbitrary point in the audio file.  Hence, when the video
message is played back, the video will be out of sync with the audio - 
both are played back starting at the same time, but the soft phone didn't
start sending the video at the beginning of the audio.

The solution to this would be to store the audio/video as a single file
in a media container (such as Matroska).  Not only does this solve the
audio/video sync issue, but now you don't need to store more then a single
file in a storage backend.  Unfortunately, this is an extremely non-trivial
effort.

--
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] No audio playing back voicemail from odbc

[Support]

On Saturday 28 July 2012 9:19:32 am Matthew Jordan wrote:
> - Original Message -
> 
> > I'm trying to get my voicemail messages stored in a mysql database
> > via odbc. Most of it is working, except, when I play my
> > voicemail messages, I don't hear anything.
> 
> Your message has a duration of 0.  So while a file was created, it does
> not apparently have an audio length.

I didn't catch that!  However, when I reconfigured for filesystem storage I had 
the same problem, so I'm not thinking is has anything to do with ODBC any 
more.  I've loaded all of the format and codec modules and still can't get 
this working.

CLI> core show translation paths slin
--- Translation paths SRC Codec "slin" sample rate 8000 ---
slin   To g723  : No Translation Path   
  
slin   To gsm   : (slin)->(gsm) 
  
slin   To ulaw  : (slin)->(ulaw)   

I'm using the ulaw audio codec and wav for storage, so this SHOULD work


> What is your voicemail.conf configuration?  Do you have a minduration
> set?
> 

I don't, but this configuration worked before I upgraded from 1.6.x to 10.x.  I 
should have mentioned that this was part of an upgrade, but it was late, and I 
was tired.

So, is there something I'm missing?

> Asterisk stores video voicemail messages as both a video file and an
> audio file.  Since the ODBC/IMAP backends only store a single file
> format, you will either end up storing only the video, or only the
> audio, which is probably not what you want.
> 
> File storage is the only mechanism to have video voicemail (with audio)
> at this time.

Is there any interest in fixing this situation?  It doesn't seem like it would 
be too difficult.  I wouldn't mind helping if there is an effort already 
underway.

-- 

Take care and have fun,
Mike Diehl.

-- 

Mike Diehl.

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Re: [asterisk-users] No audio playing back voicemail from odbc

[Matthew Jordan]

- Original Message - 

> From: "Support" 
> To: asterisk-users@lists.digium.com
> Sent: Saturday, July 28, 2012 4:43:55 AM
> Subject: [asterisk-users] No audio playing back voicemail from odbc

> Hi all,

> I'm trying to get my voicemail messages stored in a mysql database
> via odbc. Most of it is working, except, when I play my
> voicemail messages, I don't hear anything.

> I can confirm that the messages are getting stored in the database:

> select
> uniqueid,msgnum,dir,context,macrocontext,callerid,origtime,duration,mailboxuser,mailboxcontext,flag,`read`
> from voicemessages;
> +--++-+---+--+++--+-++--+--+
> | uniqueid | msgnum | dir | context | macrocontext | callerid |
> | origtime | duration | mailboxuser | mailboxcontext | flag | read |
> +--++-+---+--+++--+-++--+--+
> | 7 | 0 | /var/spool/asterisk/voicemail/diehlnet/7001/Old | customers
> | | | 17442025-1 | 1343462922 | 0 | 7001 | diehlnet | | 0 |
> +--++-+---+--+++--+-++--+--+

Your message has a duration of 0.  So while a file was created, it does
not apparently have an audio length.

What is your voicemail.conf configuration?  Do you have a minduration
set?

> select length(recording) from voicemessages;
> +---+
> | length(recording) |
> +---+
> | 52204 |
> +---+

> This is what the console displays during message playback:

> [Jul 28 02:28:35] == Parsing
> '/var/spool/asterisk/voicemail/diehlnet/7001/Old/msg.txt': [Jul
> 28 02:28:35] == Found
> [Jul 28 02:28:35] --  Playing
> 'vm-message.ulaw' (language 'en')
> [Jul 28 02:28:36] --  Playing
> 'vm-unknown-caller.ulaw' (language 'en')
> [Jul 28 02:28:38] --  Playing
> '/var/spool/asterisk/voicemail/diehlnet/7001/Old/msg.slin'
> (language 'en')

> Of course mg.txt and msg.slin don't exist. I'm assuming
> Asterisk creates them and deletes them?

When Asterisk plays a message from a non-file message storage backend,
it extracts from the backend and stores it locally on the file system.
It then plays the file from there.

When it has finished, it does delete it.  So yes, the log messages
are correct - it is playing the file from that location.

> Also, on a side note, ODBC storage and video voicemail aren't going
> to work together, right?

Asterisk stores video voicemail messages as both a video file and an
audio file.  Since the ODBC/IMAP backends only store a single file
format, you will either end up storing only the video, or only the
audio, which is probably not what you want.

File storage is the only mechanism to have video voicemail (with audio)
at this time.


--
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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[asterisk-users] No audio playing back voicemail from odbc

[Support]

Hi all,

I'm trying to get my voicemail messages stored in a mysql database via odbc.  
Most of it is working, except, when I play my 
voicemail messages, I don't hear anything.

I can confirm that the messages are getting stored in the database:

select 
uniqueid,msgnum,dir,context,macrocontext,callerid,origtime,duration,mailboxuser,mailboxcontext,flag,`read`
 
from voicemessages;
+--++-+---+--+++--+-++--+--+
| uniqueid | msgnum | dir | 
context   | macrocontext | callerid   | origtime   | duration | 
mailboxuser | mailboxcontext | flag | read |
+--++-+---+--+++--+-++--+--+
|7 |  0 | /var/spool/asterisk/voicemail/diehlnet/7001/Old | 
customers |  | 17442025-1 | 1343462922 | 0| 7001
| diehlnet   |  |0 |
+--++-+---+--+++--+-++--+--+

select length(recording) from voicemessages;
+---+
| length(recording) |
+---+
| 52204 |
+---+

This is what the console displays during message playback:

[Jul 28 02:28:35]   == Parsing 
'/var/spool/asterisk/voicemail/diehlnet/7001/Old/msg.txt': [Jul 28 
02:28:35]   == Found
[Jul 28 02:28:35] --  Playing 'vm-
message.ulaw' (language 'en')
[Jul 28 02:28:36] --  Playing 'vm-unknown-
caller.ulaw' (language 'en')
[Jul 28 02:28:38] --  Playing 
'/var/spool/asterisk/voicemail/diehlnet/7001/Old/msg.slin' (language 'en')

Of course mg.txt and msg.slin don't exist.  I'm assuming Asterisk 
creates them and deletes them?

But I don't hear anything.  Any ideas?  Asterisk version 10.6.1 

Also, on a side note, ODBC storage and video voicemail aren't going to work 
together, right?

-- 

Take care and have fun,
Mike Diehl.
-- 

Mike Diehl.
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Re: [asterisk-users] NO AUDIO

[Thiago Coutinho]

On Tue, Jul 10, 2012 at 10:45 AM, Daviramos Roussenq Fortunato
 wrote:
> I have a server running at more than two years with Asterisk 1.6, and began
> presenting problem seedlings links in external SIP extensions on some links.
>
> By doing "rtp set debug on" discovered the problem, he is trying to deliver
> the audio directly to internal IP Extension. And sometimes shown correctly
> on the external IP, where this time the link works correctly. These
> extensions are at Stake sip configured with nat = yes and = no canreivinte.
>
> In the "sip show settings" I have my ip correctly list "externip: MY_IP:
> 5060" Server is not behind NAT, the IP is directly on it, it has only one
> network card, the SIM remote extensions are behind NAT.
>
> What may be occurring in some links for it to work correctly and not others?

I don't know if it will resolve, but try add the "localnet" option in sip.conf.

-- 
thiagoc

"O povo não deveria temer o governo. O governo é quem deveria temer o povo."
V de Vingança

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[asterisk-users] NO AUDIO

[Daviramos Roussenq Fortunato]

Hi,


I have a server running at more than two years with Asterisk 1.6, and began
presenting problem seedlings links in external SIP extensions on some links.

By doing "rtp set debug on" discovered the problem, he is trying to deliver
the audio directly to internal IP Extension. And sometimes shown correctly
on the external IP, where this time the link works correctly. These
extensions are at Stake sip configured with nat = yes and = no canreivinte.

In the "sip show settings" I have my ip correctly list "externip: MY_IP:
5060" Server is not behind NAT, the IP is directly on it, it has only one
network card, the SIM remote extensions are behind NAT.

What may be occurring in some links for it to work correctly and not others?

-- 
Atenciosamente
Daviramos Roussenq Fortunato
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Re: [asterisk-users] No audio available on SIP/172.16.129.13:5060-00000001??

[shalu dhamija]

Hello, 

Actually I have changed asterisk in such a way that any call that comes onto 
asterisk server will go into the voicemail() application for that user. 

I am sending the media through SIPp by putting the following action in scenario 
file: 





 
   
     
   
     
   



Regards, 

Shalu 


Date: Wed, 11 Jan 2012 10:59:33 +0530 

From: virendra bhati  

Subject: Re: [asterisk-users] No audio available on 

  SIP/172.16.129.13:5060-0001?? 

To: Asterisk Users Mailing List - Non-Commercial Discussion 

   

Message-ID: 

   

Content-Type: text/plain; charset="iso-8859-1" 





Hi Shalu, 



  

How you are invoking call in dialplan. it's completely depends on that. 

And error show that no voice is there for store in voicemail . 



  

On Wed, Jan 11, 2012 at 10:05 AM, shalu dhamija < 

shalu.dham...@rancoretech.com > wrote: 



  

> Hello, 

> 

  

> 

  

> 

  

> I am trying to run load on asterisk server(version 1.8.7.1) for the 

> voicemail() application using SIPp tool. I am just running sipp at call 

> rate of 1 cps with the following command: 

> 

  

> 

  

> 

  

> ./sipp -m 9000 -r 1 -rp 1000 -d 45 -max_socket 65536 -sf 

> uac_msg_deposit.xml -i 172.16.129.13 -s 1 172.16.129.14 --trace_err 

> 

  

> 

  

> 

  

> I am trying to deposit 9000 messages in the mailbox of user 1 (given by 

> the -s option) but the following warning is coming on the asterisk server 

> due to which the message does not get deposited into the users mailbox: 

> 

  

> 

  

> 

  

> No audio available on SIP/172.16.129.13:5060-0001?? 

> 

  

> 

  

> 

  

> I have set rtpstart=6000 and rtpend=2 in rtp.conf. 

> 

  

> 

  

> 

  

> 

  

> 

  

> Can someone please let me know how to avoid these kind of warnings. 

> 

  

> 

  

> 

  

> Thanks. 

> 

  

> 

  

> 

  

> Shalu 

> 

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Re: [asterisk-users] No audio available on SIP/172.16.129.13:5060-00000001??

[virendra bhati]

Hi Shalu,

How you are invoking call in dialplan. it's completely depends on that.
And error show that no voice is there for store in voicemail .

On Wed, Jan 11, 2012 at 10:05 AM, shalu dhamija <
shalu.dham...@rancoretech.com> wrote:

> Hello,
>
>
>
> I am trying to run load on asterisk server(version 1.8.7.1) for the
> voicemail() application using SIPp tool. I am just running sipp at call
> rate of 1 cps with the following command:
>
>
>
> ./sipp -m 9000 -r 1 -rp 1000 -d 45 -max_socket 65536 -sf
> uac_msg_deposit.xml -i 172.16.129.13 -s 1 172.16.129.14 --trace_err
>
>
>
> I am trying to deposit 9000 messages in the mailbox of user 1 (given by
> the -s option) but the following warning is coming on the asterisk server
> due to which the message does not get deposited into the users mailbox:
>
>
>
> No audio available on SIP/172.16.129.13:5060-0001??
>
>
>
> I have set rtpstart=6000 and rtpend=2 in rtp.conf.
>
>
>
>
>
> Can someone please let me know how to avoid these kind of warnings.
>
>
>
> Thanks.
>
>
>
> Shalu
>
>
>
>
>
>
>
> Thanks and Regards,
> Shalu Dhamija
> Rancore Technologies(P) Ltd.
> Gurgaon
> Ph : 0124-4200691
> +91-9910995356(M)
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 

Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer
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[asterisk-users] No audio available on SIP/172.16.129.13:5060-00000001??

[shalu dhamija]

Hello, 



I am trying to run load on asterisk server(version 1.8.7.1) for the voicemail() 
application using SIPp tool. I am just running sipp at call rate of 1 cps with 
the following command: 



./sipp -m 9000 -r 1 -rp 1000 -d 45 -max_socket 65536 -sf uac_msg_deposit.xml -i 
172.16.129.13 -s 1 172.16.129.14 --trace_err 



I am trying to deposit 9000 messages in the mailbox of user 1 (given by the -s 
option) but the following  warning is coming on the asterisk server due to 
which the message does not get deposited into the users mailbox: 

  

No audio available on SIP/172.16.129.13:5060-0001?? 



I have set rtpstart=6000 and rtpend=2 in rtp.conf. 





Can someone please let me know how to avoid these kind of warnings. 



Thanks. 



Shalu 







Thanks and Regards, 
Shalu Dhamija 
Rancore Technologies(P) Ltd. 
Gurgaon 
Ph : 0124-4200691 
+91-9910995356(M) 
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[asterisk-users] no audio using g729A for Cisco AS5300 sip peer

[Roi Stork]

Hi,

We need help in enabling g729a codec for our SIP peer that's using Cisco
AS5300.
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.
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Re: [asterisk-users] No Audio after attended tranfer

[Alex Vishnev]

No, that looks like a separate issue. Mine is a 100% repeatable and the 
asterisk does not lock up. SIP and RTP on other sessions are still going. in my 
cases this is the exchange I see

Asterisk
  Service Provider
INVITE (initial Invite to Service Provider with Outbound number) --->
<--200 OK
<-INVITE (put session on hold)
-->200OK 
<--ACK
<->RTP
<<-INVITE (no SDP) -- First transfer complete
-->200OK (SDP)
<--ACK
<->RTP
<<-INVITE (no SDP) -- Second Transfer
-->200OK (SDP)
<--ACK (SDP)
<--RTP

On Jul 19, 2011, at 3:41 AM, Stefan Schmidt wrote:

> Am 18.07.11 16:15, schrieb Alex Vishnev:
>> I am wondering if anyone hit this case yet. I am using 1.6.2.19 and doing an 
>> attended transfer. The transfer is going to an outbound number (normally AA 
>> on another IP PBX). the audio on the first transfer is fine. But if the user 
>> requests a transfer from AA to another department, I loose audio from 
>> Asterisk to the 2nd transfer. Based on the review of SIP packets, the second 
>> transfer issues ACK+SDP. Anyone have experience with that? it looks like 
>> ACK+SDP is not being handled properly by asterisk. I searched thru JIRA 
>> cases, but did not find anything like that. Any help would be appreciated.
>> --
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> Hello,
> 
> maybe this is the problem you have:
> 
> https://issues.asterisk.org/jira/browse/ASTERISK-18136
> 
> best regards
> 
> Stefan
> 
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Re: [asterisk-users] No Audio after attended tranfer

[Stefan Schmidt]

Am 18.07.11 16:15, schrieb Alex Vishnev:
> I am wondering if anyone hit this case yet. I am using 1.6.2.19 and doing an 
> attended transfer. The transfer is going to an outbound number (normally AA 
> on another IP PBX). the audio on the first transfer is fine. But if the user 
> requests a transfer from AA to another department, I loose audio from 
> Asterisk to the 2nd transfer. Based on the review of SIP packets, the second 
> transfer issues ACK+SDP. Anyone have experience with that? it looks like 
> ACK+SDP is not being handled properly by asterisk. I searched thru JIRA 
> cases, but did not find anything like that. Any help would be appreciated.
> --
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Hello,

maybe this is the problem you have:

https://issues.asterisk.org/jira/browse/ASTERISK-18136

best regards

Stefan

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[asterisk-users] No Audio after attended tranfer

[Alex Vishnev]

I am wondering if anyone hit this case yet. I am using 1.6.2.19 and doing an 
attended transfer. The transfer is going to an outbound number (normally AA on 
another IP PBX). the audio on the first transfer is fine. But if the user 
requests a transfer from AA to another department, I loose audio from Asterisk 
to the 2nd transfer. Based on the review of SIP packets, the second transfer 
issues ACK+SDP. Anyone have experience with that? it looks like ACK+SDP is not 
being handled properly by asterisk. I searched thru JIRA cases, but did not 
find anything like that. Any help would be appreciated.
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Re: [asterisk-users] No audio after a reinvite changing codec ----> with SIP DEBUG!!

[Matteo Campana]

On Fri, Jul 1, 2011 at 12:05 PM, Larry Moore  wrote:

> **
> On 28/06/2011 6:59 PM, Matteo Campana wrote:
>
>
>
>  Hi Larry,
> I have the SIP debug taken from asterisk.
> In this debug: 1.2.3.4 ---> IP SIP PROXY
>  5.6.7.8 ---> IP UAC (Linksys SPA 962)
>  9.10.11.12 ---> IP ASTERISK to connect to the
> provider
>  13.14.15.16 --> IP PROVIDER
>  17.18.19.20 --> IP ASTERISK
>
>
>  The SIP debug is available at this link: http://pastebin.com/9DrFDmeC
>
>
>
> You mention you have an SPA962, I expect the configuration will be the same
> if not similar to an SPA942. It would be worth checking what your "Symmetric
> RTP" setting is, you can find it listed in the RTP Parameters section under
> the SIP section of your phone http://
> /admin/advanced.
>
> If it is set to "no" set it to "yes".
>
> Larry.
>
>
Hy Larry,
I have tested with "Symmetric RTP = yes" in SPA962, but with same results.

Matteo
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Re: [asterisk-users] No audio after a reinvite changing codec ----> with SIP DEBUG!!

[Larry Moore]

On 28/06/2011 6:59 PM, Matteo Campana wrote:



Hi Larry,
I have the SIP debug taken from asterisk.
In this debug: 1.2.3.4 ---> IP SIP PROXY
 5.6.7.8 ---> IP UAC (Linksys SPA 962)
 9.10.11.12 ---> IP ASTERISK to connect to the 
provider

 13.14.15.16 --> IP PROVIDER
 17.18.19.20 --> IP ASTERISK


The SIP debug is available at this link: http://pastebin.com/9DrFDmeC




You mention you have an SPA962, I expect the configuration will be the 
same if not similar to an SPA942. It would be worth checking what your 
"Symmetric RTP" setting is, you can find it listed in the RTP Parameters 
section under the SIP section of your phone 
http:///admin/advanced.


If it is set to "no" set it to "yes".

Larry.
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Re: [asterisk-users] No audio format found to offer.

[Eric Wieling]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> Ernie Dunbar
> Sent: Wednesday, June 29, 2011 6:34 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] No audio format found to offer.
>
> Quoting Carlos Chavez :
>
>
> > The disallow line must be set before any allow line.
> >
> > Since Asterisk has no official G723 support you should
> not even be
> > trying to use that.
>
> That's fantastic. I'll tell that to our SIP trunk provider right away.
>
> > Do you have the G.279 codec and license  installed in your system?
> > Remember that G.729 is not included in Asterisk (as a
> > codec) so it only works in passthru.
>
> So G.729 will only work for this trunk if the customer's ATA
> is using it too?

Assuming Asterisk does not have to transcode yes.  Transcoding is required to 
play Asterisk sound files (if g729 versions are not installed), the T/t/W/w 
option to Dial, ChanSpy, etc.  "Pass thru" may sound cool, but it seldom works 
well in the real world.  Spend the money on a G729 license from Digium 
($10/channel) and save yourself problems.

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Re: [asterisk-users] No audio format found to offer.

[Ernie Dunbar]

Quoting Carlos Chavez :



The disallow line must be set before any allow line.

Since Asterisk has no official G723 support you should not even be
trying to use that.


That's fantastic. I'll tell that to our SIP trunk provider right away.


Do you have the G.279 codec and license  installed
in your system?  Remember that G.729 is not included in Asterisk (as a
codec) so it only works in passthru.


So G.729 will only work for this trunk if the customer's ATA is using it too?


You need to purchase some licenses
and install the codec for it to work.


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Director de Tecnología
+52-55-91169161 ext 2001






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Re: [asterisk-users] No audio format found to offer.

[Carlos Chavez]

On Wed, 2011-06-29 at 18:12 -0400, Alex Balashov wrote:
> Perhaps do this instead?
> 
>allow=g723
>allow=g729
>disallow=all
> 
> On 06/29/2011 05:57 PM, Ernie Dunbar wrote:
> 
> > This *should* be something that's easy to fix, but apparently I'm not
> > doing something right.
> >
> > Our SIP long distance provider is telling us to only use formats G.723
> > and G.729, so I've set up their trunk configuration in sip.conf as such:
> >
> > [t564]
> > type=friend
> > host=XXX.XX.56.4
> > context=default
> > disallow=all
> > allow=g723
> > allow=g729
> >
> > However, the Dial application gives the following error:
> >
> > -- AGI Script Executing Application: (DIAL) Options:
> > (SIP/t564/1XX4332,,HR)
> > == Using SIP RTP CoS mark 5
> > [Jun 29 13:37:37] WARNING[16693]: chan_sip.c:5518 sip_call: No audio
> > format found to offer. Cancelling call to 1XX4332
> > -- Couldn't call t564/1XX332
> > == Everyone is busy/congested at this time (0:0/0/0)
> >
> > I've checked to ensure that both formats are loaded into Asterisk:
> >
> > voip2*CLI> module show like 729
> > Module Description Use Count
> > format_g729.so Raw G729 data 0
> > 1 modules loaded
> > voip2*CLI> module show like 723
> > Module Description Use Count
> > format_g723.so G.723.1 Simple Timestamp File Format 0
> > 1 modules loaded
> >
> > So I'm at a bit of a loss as to why Asterisk is complaining that there's
> > no audio format found to offer.
> >
The disallow line must be set before any allow line.

Since Asterisk has no official G723 support you should not even be
trying to use that.  Do you have the G.279 codec and license  installed
in your system?  Remember that G.729 is not included in Asterisk (as a
codec) so it only works in passthru.  You need to purchase some licenses
and install the codec for it to work.


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Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] No audio format found to offer.

[Alex Balashov]

Perhaps do this instead?

  allow=g723
  allow=g729
  disallow=all

On 06/29/2011 05:57 PM, Ernie Dunbar wrote:


This *should* be something that's easy to fix, but apparently I'm not
doing something right.

Our SIP long distance provider is telling us to only use formats G.723
and G.729, so I've set up their trunk configuration in sip.conf as such:

[t564]
type=friend
host=XXX.XX.56.4
context=default
disallow=all
allow=g723
allow=g729

However, the Dial application gives the following error:

-- AGI Script Executing Application: (DIAL) Options:
(SIP/t564/1XX4332,,HR)
== Using SIP RTP CoS mark 5
[Jun 29 13:37:37] WARNING[16693]: chan_sip.c:5518 sip_call: No audio
format found to offer. Cancelling call to 1XX4332
-- Couldn't call t564/1XX332
== Everyone is busy/congested at this time (0:0/0/0)

I've checked to ensure that both formats are loaded into Asterisk:

voip2*CLI> module show like 729
Module Description Use Count
format_g729.so Raw G729 data 0
1 modules loaded
voip2*CLI> module show like 723
Module Description Use Count
format_g723.so G.723.1 Simple Timestamp File Format 0
1 modules loaded

So I'm at a bit of a loss as to why Asterisk is complaining that there's
no audio format found to offer.


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--
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Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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[asterisk-users] No audio format found to offer.

[Ernie Dunbar]

This *should* be something that's easy to fix, but apparently I'm not  
doing something right.


Our SIP long distance provider is telling us to only use formats G.723  
and G.729, so I've set up their trunk configuration in sip.conf as such:


[t564]
type=friend
host=XXX.XX.56.4
context=default
disallow=all
allow=g723
allow=g729

However, the Dial application gives the following error:

-- AGI Script Executing Application: (DIAL) Options:  
(SIP/t564/1XX4332,,HR)

  == Using SIP RTP CoS mark 5
[Jun 29 13:37:37] WARNING[16693]: chan_sip.c:5518 sip_call: No audio  
format found to offer. Cancelling call to 1XX4332

-- Couldn't call t564/1XX332
  == Everyone is busy/congested at this time (0:0/0/0)

I've checked to ensure that both formats are loaded into Asterisk:

voip2*CLI> module show like 729
Module Description  
 Use Count

format_g729.so Raw G729 data0
1 modules loaded
voip2*CLI> module show like 723
Module Description  
 Use Count

format_g723.so G.723.1 Simple Timestamp File Format 0
1 modules loaded

So I'm at a bit of a loss as to why Asterisk is complaining that  
there's no audio format found to offer.



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Re: [asterisk-users] No audio after a reinvite changing codec ----> with SIP DEBUG!!

[Matteo Campana]

On Sat, Jun 18, 2011 at 6:40 AM, Larry Moore  wrote:

> On 18/06/2011 5:36 AM, Matteo Campana wrote:
>
>>
>> Inviato da iPhone
>>
>> Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling
>>  ha scritto:
>>
>>  We experience the same thing.  The solution we use is to not change
>>> codecs in the middle of a call.   I assumed it was an issue with our
>>> upstream.
>>>
>>
>> Hi Eric,
>> this behavior  is an asterisk bug or asterisk can never change the codec
>> "on the fly"?
>>
>>
>> Thanks,
>> Matteo
>>
>>
> The problem reported occurs after a fax tone is detected, one might expect
> T.38 or G711 to be used to handle the fax, not G729!
>
> There is no configuration file information for each of the nodes/peers, no
> debug of each peer involved  nor a trace of the packets hence no one will
> have ideas!
>
> Larry.



Hi Larry,
I have the SIP debug taken from asterisk.
In this debug: 1.2.3.4 ---> IP SIP PROXY
 5.6.7.8 ---> IP UAC (Linksys SPA 962)
 9.10.11.12 ---> IP ASTERISK to connect to the
provider
 13.14.15.16 --> IP PROVIDER
 17.18.19.20 --> IP ASTERISK


The SIP debug is available at this link: http://pastebin.com/9DrFDmeC


Thanks in advance,
Matteo










>
>
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Re: [asterisk-users] No audio after a reinvite changing codec

[Matteo Campana]

On Mon, Jun 20, 2011 at 11:58 PM, Matteo Campana
wrote:

>
>
> Inviato da iPhone
>
> Il giorno 18/giu/2011, alle ore 06:40, Larry Moore 
> ha scritto:
>
> > On 18/06/2011 5:36 AM, Matteo Campana wrote:
> >>
> >> Inviato da iPhone
> >>
> >> Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling
>  ha scritto:
> >>
> >>> We experience the same thing.  The solution we use is to not change
> codecs in the middle of a call.   I assumed it was an issue with our
> upstream.
> >>
> >> Hi Eric,
> >> this behavior  is an asterisk bug or asterisk can never change the codec
> "on the fly"?
> >>
> >>
> >> Thanks,
> >> Matteo
> >>
> >
> > The problem reported occurs after a fax tone is detected, one might
> expect T.38 or G711 to be used to handle the fax, not G729!
> >
> > There is no configuration file information for each of the nodes/peers,
> no debug of each peer involved  nor a trace of the packets hence no one will
> have ideas!
> >
> > Larry.
> >
>

 Hi,
I'm out of the office this week, next Monday I will send the debug to the
list.
However I think It's strange asterisk behavior: it says 200 OK after a
re-invite by the provider, but stops to send rtp.

Regards,

Matteo
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Re: [asterisk-users] No audio after a reinvite changing codec

[Matteo Campana]

Inviato da iPhone

Il giorno 18/giu/2011, alle ore 06:40, Larry Moore  ha 
scritto:

> On 18/06/2011 5:36 AM, Matteo Campana wrote:
>> 
>> Inviato da iPhone
>> 
>> Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling  ha 
>> scritto:
>> 
>>> We experience the same thing.  The solution we use is to not change codecs 
>>> in the middle of a call.   I assumed it was an issue with our upstream.
>> 
>> Hi Eric,
>> this behavior  is an asterisk bug or asterisk can never change the codec "on 
>> the fly"?
>> 
>> 
>> Thanks,
>> Matteo
>> 
> 
> The problem reported occurs after a fax tone is detected, one might expect 
> T.38 or G711 to be used to handle the fax, not G729!
> 
> There is no configuration file information for each of the nodes/peers, no 
> debug of each peer involved  nor a trace of the packets hence no one will 
> have ideas!
> 
> Larry.
> 


Hi,
I'm out of the office this week, next Monday I will send the debug to the list.

However I think It's strange asterisk behavior: it says 200 OK after a 
re-invite by the provider, but stops to send rtp.


Regards,
Matteo
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Re: [asterisk-users] No audio after a reinvite changing codec

[Larry Moore]

On 18/06/2011 5:36 AM, Matteo Campana wrote:


Inviato da iPhone

Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling  ha 
scritto:


We experience the same thing.  The solution we use is to not change codecs in 
the middle of a call.   I assumed it was an issue with our upstream.


Hi Eric,
this behavior  is an asterisk bug or asterisk can never change the codec "on the 
fly"?


Thanks,
Matteo



The problem reported occurs after a fax tone is detected, one might 
expect T.38 or G711 to be used to handle the fax, not G729!


There is no configuration file information for each of the nodes/peers, 
no debug of each peer involved  nor a trace of the packets hence no one 
will have ideas!


Larry.

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Re: [asterisk-users] No audio after a reinvite changing codec

[Eric Wieling]

I don't know.

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> Matteo Campana
> Sent: Friday, June 17, 2011 5:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] No audio after a reinvite changing codec
>
>
>
> Inviato da iPhone
>
> Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling
>  ha scritto:
>
> >
> > We experience the same thing.  The solution we use is to
> not change codecs in the middle of a call.   I assumed it was
> an issue with our upstream.
>
>
> Hi Eric,
> this behavior  is an asterisk bug or asterisk can never
> change the codec "on the fly"?
>
>
> Thanks,
> Matteo
>
>
>
>
> >
> >> -Original Message-
> >> From: asterisk-users-boun...@lists.digium.com
> >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
> Of Larry
> >> Moore
> >> Sent: Thursday, June 16, 2011 10:32 AM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [asterisk-users] No audio after a reinvite changing
> >> codec
> >>
> >> On 15/06/2011 8:15 PM, Matteo Campana wrote:
> >>
> >>  HI list,
> >>  no idea?? :)
> >>
> >>
> >>
> >> There not much substance in the information provided for an
> >> assessment to be made.
> >>
> >> I would suggest you capture the network traffic between UAC
> >> (g711) & Asterisk UAS ensuring the snap length is large enough to
> >> capture the whole packet and do the same with traffic between
> >> Asterisk UAC & Provider then use Wireshark and its
> telephony feature
> >> to analyse VoIP calls, check the flows, you may discover
> the problem
> >> this way!
> >>
> >> Larry.
> >>
> >>
> >>
> >>  M.
> >>
> >>
> >>  On Mon, Jun 13, 2011 at 6:55 PM, Matteo Campana
> >>  wrote:
> >>
> >>
> >>  Hi all,
> >>  we have a problem with a reinvite sent by our SIP
> >> provider to change audio codec due to the recognition of a
> fax tone.
> >>  After that the SIP call session has been established
> >> (INVITE and 200 OK) we have the following codec
> >> situation:
> >>
> >>  UAC
> >> ASTERISK UAS | ASTERISK UAC  PROVIDER
> >>  g711  <-->
> >> g711  |   g729 <--->  g729
> >>  rtp
> >>   rtp
> >>
> >>  After a while, we have the reinvite sent by the SIP
> >> provider with g711 in the SDP.
> >>  So asterisk need to change audio codec from
> >> g729 to g711 and correctly we see on debug the following line:
> >>  "Oooh, we need to change our audio formats since our
> >> peer supports only g729" and asterisk send back 200 OK to the
> >> provider.
> >>  At this point we have one way rtp audio:
> >>
> >>  UAC
> >> ASTERISK UAS | ASTERISK UAC  PROVIDER
> >>  g711  -->
> >> g711  |   g711 --->  g711
> >>  rtp
> >>   rtp
> >>
> >>  So the problem is that UAC does not hear audio at all.
> >>  Any idea?
> >>
> >>  (Asterisk version: 1.4.33.1).
> >>
> >>  Thanks in advance,
> >>  Matteo
> >>
> >>
> >>
> >>
> >>  --
> >>
> >>
> _
> >>  -- Bandwidth and Colocation Provided by
> >> http://www.api-digital.com --
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> webinar every
> >> Thurs:
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> >> http://lists.digium.

Re: [asterisk-users] No audio after a reinvite changing codec

[Matteo Campana]

Inviato da iPhone

Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling  ha 
scritto:

> 
> We experience the same thing.  The solution we use is to not change codecs in 
> the middle of a call.   I assumed it was an issue with our upstream.


Hi Eric,
this behavior  is an asterisk bug or asterisk can never change the codec "on 
the fly"?


Thanks,
Matteo




> 
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
>> Larry Moore
>> Sent: Thursday, June 16, 2011 10:32 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] No audio after a reinvite changing codec
>> 
>> On 15/06/2011 8:15 PM, Matteo Campana wrote:
>> 
>>  HI list,
>>  no idea?? :)
>> 
>> 
>> 
>> There not much substance in the information provided for an
>> assessment to be made.
>> 
>> I would suggest you capture the network traffic between UAC
>> (g711) & Asterisk UAS ensuring the snap length is large
>> enough to capture the whole packet and do the same with
>> traffic between Asterisk UAC & Provider then use Wireshark
>> and its telephony feature to analyse VoIP calls, check the
>> flows, you may discover the problem this way!
>> 
>> Larry.
>> 
>> 
>> 
>>  M.
>> 
>> 
>>  On Mon, Jun 13, 2011 at 6:55 PM, Matteo Campana
>>  wrote:
>> 
>> 
>>  Hi all,
>>  we have a problem with a reinvite sent by our
>> SIP provider to change audio codec due to the recognition of
>> a fax tone.
>>  After that the SIP call session has been
>> established (INVITE and 200 OK) we have the following codec
>> situation:
>> 
>>  UAC
>> ASTERISK UAS | ASTERISK UAC  PROVIDER
>>  g711  <-->
>> g711  |   g729 <--->  g729
>>  rtp
>>   rtp
>> 
>>  After a while, we have the reinvite sent by the
>> SIP provider with g711 in the SDP.
>>  So asterisk need to change audio codec from
>> g729 to g711 and correctly we see on debug the following line:
>>  "Oooh, we need to change our audio formats
>> since our peer supports only g729" and asterisk send back 200
>> OK to the provider.
>>  At this point we have one way rtp audio:
>> 
>>  UAC
>> ASTERISK UAS | ASTERISK UAC  PROVIDER
>>  g711  -->
>> g711  |   g711 --->  g711
>>  rtp
>>   rtp
>> 
>>  So the problem is that UAC does not hear audio at all.
>>  Any idea?
>> 
>>  (Asterisk version: 1.4.33.1).
>> 
>>  Thanks in advance,
>>  Matteo
>> 
>> 
>> 
>> 
>>  --
>> 
>> _
>>  -- Bandwidth and Colocation Provided by
>> http://www.api-digital.com --
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>> webinar every Thurs:
>> http://www.asterisk.org/hello
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>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> 
>> 
>> 
> 
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Re: [asterisk-users] No audio after a reinvite changing codec

[Eric Wieling]

We experience the same thing.  The solution we use is to not change codecs in 
the middle of a call.   I assumed it was an issue with our upstream.

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> Larry Moore
> Sent: Thursday, June 16, 2011 10:32 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] No audio after a reinvite changing codec
>
> On 15/06/2011 8:15 PM, Matteo Campana wrote:
>
>   HI list,
>   no idea?? :)
>
>
>
> There not much substance in the information provided for an
> assessment to be made.
>
> I would suggest you capture the network traffic between UAC
> (g711) & Asterisk UAS ensuring the snap length is large
> enough to capture the whole packet and do the same with
> traffic between Asterisk UAC & Provider then use Wireshark
> and its telephony feature to analyse VoIP calls, check the
> flows, you may discover the problem this way!
>
> Larry.
>
>
>
>   M.
>
>
>   On Mon, Jun 13, 2011 at 6:55 PM, Matteo Campana
>  wrote:
>
>
>   Hi all,
>   we have a problem with a reinvite sent by our
> SIP provider to change audio codec due to the recognition of
> a fax tone.
>   After that the SIP call session has been
> established (INVITE and 200 OK) we have the following codec
> situation:
>
>   UAC
> ASTERISK UAS | ASTERISK UAC  PROVIDER
>   g711  <-->
> g711  |   g729 <--->  g729
>   rtp
>rtp
>
>   After a while, we have the reinvite sent by the
> SIP provider with g711 in the SDP.
>   So asterisk need to change audio codec from
> g729 to g711 and correctly we see on debug the following line:
>   "Oooh, we need to change our audio formats
> since our peer supports only g729" and asterisk send back 200
> OK to the provider.
>   At this point we have one way rtp audio:
>
>   UAC
> ASTERISK UAS | ASTERISK UAC  PROVIDER
>   g711  -->
> g711  |   g711 --->  g711
>   rtp
>rtp
>
>   So the problem is that UAC does not hear audio at all.
>   Any idea?
>
>   (Asterisk version: 1.4.33.1).
>
>   Thanks in advance,
>   Matteo
>
>
>
>
>   --
>
> _
>   -- Bandwidth and Colocation Provided by
> http://www.api-digital.com --
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> webinar every Thurs:
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>
>
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Re: [asterisk-users] No audio after a reinvite changing codec

[Larry Moore]

On 15/06/2011 8:15 PM, Matteo Campana wrote:

HI list,
no idea?? :)



There not much substance in the information provided for an assessment 
to be made.


I would suggest you capture the network traffic between UAC (g711) & 
Asterisk UAS ensuring the snap length is large enough to capture the 
whole packet and do the same with traffic between Asterisk UAC & 
Provider then use Wireshark and its telephony feature to analyse VoIP 
calls, check the flows, you may discover the problem this way!


Larry.


M.

On Mon, Jun 13, 2011 at 6:55 PM, Matteo Campana 
mailto:matteo.camp...@gmail.com>> wrote:


Hi all,
we have a problem with a reinvite sent by our SIP provider to
change audio codec due to the recognition of a fax tone.
After that the SIP call session has been established (INVITE and
200 OK) we have the following codec situation:

UACASTERISK UAS | ASTERISK
UAC  PROVIDER
g711 <-->   g711  |   g729
<--->  g729
rtp   
rtp


After a while, we have the reinvite sent by the SIP provider with
g711 in the SDP.
So asterisk need to change audio codec from g729 to g711 and
correctly we see on debug the following line:
"Oooh, we need to change our audio formats since our peer supports
only g729" and asterisk send back 200 OK to the provider.
At this point we have one way rtp audio:

UACASTERISK UAS | ASTERISK
UAC  PROVIDER
g711  -->   g711  |   g711
--->  g711
rtp   
rtp


So the problem is that UAC does not hear audio at all.
Any idea?

(Asterisk version: 1.4.33.1).

Thanks in advance,
Matteo 




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Re: [asterisk-users] No audio after a reinvite changing codec

[Matteo Campana]

HI list,
no idea?? :)

M.

On Mon, Jun 13, 2011 at 6:55 PM, Matteo Campana wrote:

> Hi all,
> we have a problem with a reinvite sent by our SIP provider to change audio
> codec due to the recognition of a fax tone.
> After that the SIP call session has been established (INVITE and 200 OK) we
> have the following codec situation:
>
> UACASTERISK UAS | ASTERISK UAC
>  PROVIDER
> g711  <-->   g711  |   g729
> <--->  g729
> rtp
> rtp
>
> After a while, we have the reinvite sent by the SIP provider with g711 in
> the SDP.
> So asterisk need to change audio codec from g729 to g711 and correctly we
> see on debug the following line:
> "Oooh, we need to change our audio formats since our peer supports only
> g729" and asterisk send back 200 OK to the provider.
> At this point we have one way rtp audio:
>
> UACASTERISK UAS | ASTERISK UAC
>  PROVIDER
> g711  -->   g711  |   g711
> --->  g711
> rtp
> rtp
>
> So the problem is that UAC does not hear audio at all.
> Any idea?
>
> (Asterisk version: 1.4.33.1).
>
> Thanks in advance,
> Matteo
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[asterisk-users] No audio after a reinvite changing codec

[Matteo Campana]

Hi all,
we have a problem with a reinvite sent by our SIP provider to change audio
codec due to the recognition of a fax tone.
After that the SIP call session has been established (INVITE and 200 OK) we
have the following codec situation:

UACASTERISK UAS | ASTERISK UAC
 PROVIDER
g711  <-->   g711  |   g729
<--->  g729
rtp
rtp

After a while, we have the reinvite sent by the SIP provider with g711 in
the SDP.
So asterisk need to change audio codec from g729 to g711 and correctly we
see on debug the following line:
"Oooh, we need to change our audio formats since our peer supports only
g729" and asterisk send back 200 OK to the provider.
At this point we have one way rtp audio:

UACASTERISK UAS | ASTERISK UAC
 PROVIDER
g711  -->   g711  |   g711
--->  g711
rtp
rtp

So the problem is that UAC does not hear audio at all.
Any idea?

(Asterisk version: 1.4.33.1).

Thanks in advance,
Matteo
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Re: [asterisk-users] no audio with SIP:INFO in meetme

[Deka, Rajib IN MAA SL]

It's working now. I removed the 'm' flag from the meetme Dialplan. It was my 
mistake. Asterisk is working fine.
Exten => 100,1,MeetMe(100,dF)

Regards,
Rajib


From: Deka, Rajib IN MAA SL
Sent: Wednesday, May 11, 2011 5:35 PM
To: 'asterisk-users@lists.digium.com'
Subject: no audio with SIP:INFO in meetme

Hello List,

Asterisk is blocking audio if 'F' flag is enabled in meetme with DTMF mode 
enabled as INFO for SIP channel.
If it is a bug in asterisk or something need to be enabled in sip.conf for the 
same.

Dialplan looks like
Exten => 100,1,MeetMe(100,dmF)

Sip.conf
dtmfmode=info

Regards,
Rajib


Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com

Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com



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Re: [asterisk-users] no audio with SIP:INFO in meetme

[DHAVAL INDRODIYA]

Hi Rajib,

There is nothing like that Asterisk is blocking an audio if you use without
F it gives you and audio or not.

cheers
Dhaval

On Wed, May 11, 2011 at 5:34 PM, Deka, Rajib IN MAA SL <
rajib.d...@siemens.com> wrote:

>  Hello List,
>
>
>
> Asterisk is blocking audio if ‘F’ flag is enabled in meetme with DTMF mode
> enabled as INFO for SIP channel.
>
> If it is a bug in asterisk or something need to be enabled in sip.conf for
> the same.
>
>
>
> Dialplan looks like
>
> Exten => 100,1,MeetMe(100,dmF)
>
>
>
> Sip.conf
>
> dtmfmode=info
>
>
>
> Regards,
>
> Rajib
>
>
>
>
>
> *Rajib Deka*
>
> SIEMENS Ltd.
>
> Robert V Chandran Tower, First Floor, West Wing,
>
> #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
>
> www.siemens.com
>
>
>
> Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com
>
>
>
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[asterisk-users] no audio with SIP:INFO in meetme

[Deka, Rajib IN MAA SL]

Hello List,

Asterisk is blocking audio if 'F' flag is enabled in meetme with DTMF mode 
enabled as INFO for SIP channel.
If it is a bug in asterisk or something need to be enabled in sip.conf for the 
same.

Dialplan looks like
Exten => 100,1,MeetMe(100,dmF)

Sip.conf
dtmfmode=info

Regards,
Rajib


Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com

Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com



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Re: [asterisk-users] No audio after 15 minutes on Asterisk 1.8

[Sherwood McGowan]

Sounds like sip sessions maybe? Just the first thing that popped in my head

On Wed, Mar 9, 2011 at 11:07 AM, Carlos Chavez wrote:

>I am having a problem with calls that last more than 15 minutes.
>  After
> 15 minutes audio stops on the call.  Sometimes just ony one side will
> stop receiving audio and other both sides.  We upgraded from 1.8.2 to
> 1.8.3 and we are still experiencing this issue.
>
>I really have no idea why this happens at the 15 minute mark and I
> cannot find any setting that as a similar time frame.  Any ideas?
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
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[asterisk-users] No audio after 15 minutes on Asterisk 1.8

[Carlos Chavez]

I am having a problem with calls that last more than 15 minutes.  After
15 minutes audio stops on the call.  Sometimes just ony one side will
stop receiving audio and other both sides.  We upgraded from 1.8.2 to
1.8.3 and we are still experiencing this issue.

I really have no idea why this happens at the 15 minute mark and I
cannot find any setting that as a similar time frame.  Any ideas?

-- 
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Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] no audio on end-point when call is connected/bridged via PBX

[Robles Román , José Miguel]

> I am running Asterisk 1.8 on a cloud server.  I have had the
> same configuration running on a physical machine with a
> similar configuration.
> Thoughts?  I know I posted this yesterday but was hoping for
> some more creative comments!

If signalling works and audio don't, it probably has to do with phones behind 
NAT. It seems necessary to review the configuration of local routers.

Regards,
José Miguel

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[asterisk-users] no audio on end-point when call is connected/bridged via PBX

[Thomas Perron]

I am trying to dial through my asterisk machine from phone A to phone B.
My DID is registered properly with the SIP provider.  When I dial from
A to B it looks fine so far.
A rings B and B can pick up and the call is bridged.  However, I don't
hear any audio so therefor it is not working.
I am running Asterisk 1.8 on a cloud server.  I have had the same
configuration running on a physical machine with a similar
configuration.
Thoughts?  I know I posted this yesterday but was hoping for some more
creative comments!

Zip*CLI> sip show registry
Hostdnsmgr Username   Refresh
StateReg.Time
sip.callwithus.com:5060 N    105
Registered   Tue, 07 Dec
2010 02:36:43
1 SIP registrations.

my sip.conf
[general]
context=default
allowoverlap=no
;bindport=5060
port=5060
bindaddr=0.0.0.0
canreinvite=no ;if your asterisk box is behind a NAT ro

;register => :3...@carrier.callwithus.com
register => :3...@sip.callwithus.com

[callwithus]
type=friend
host=sip.callwithus.com
username=
secret=31
qualify=no
insecure=invite


my extensions.conf
[general]

[globals]
CONSOLE=Console/dsp ; Console interface for demo
OUTBOUNDTRUNK=SIP/callwithus

[default]
exten => s,1,Answer()
exten => s,n,Dial(SIP/callwithus/122)
exten => s,n,Wait(2)
exten => s,n,Hangup()

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Re: [asterisk-users] no audio

[Steve Edwards]

Un-top-posting...

>> On Sun, 5 Dec 2010, Thomas Perron wrote:
>>
>>> Any reason why I don't get audio on the channel after it rings and the
>>> end user picks up.
>>
>>> exten => s,n,Dial(SIP/callwithus/44,120,A,(demo-thanks))

> On Sun, Dec 5, 2010 at 8:34 PM, Steve Edwards  
> wrote:
>
>> Re-read 'core show application dial'
>>
>> Where is your prompt to option 'A' ?

On Sun, 5 Dec 2010, Thomas Perron wrote:

> negative.  no joy.
> removed the line to make is very basic.  see below.

> exten => s,1,Answer()
> exten => s,n,Wait(1)
> exten => s,n,Dial(SIP/callwithus/44)

Crank up the verbosity and debugging levels, check the codecs, etc.

Does 'sip set debug on' give any clues?

-- 
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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] no audio

[Thomas Perron]

Steve,
thanks for your note

negative.  no joy.
removed the line to make is very basic.  see below.

[globals]
CONSOLE=Console/dsp ; Console interface for demo
OUTBOUNDTRUNK=SIP/callwithus


;[general]


[default]
include => stdexten
exten => s,1,Answer()
exten => s,n,Wait(1)
exten => s,n,Dial(SIP/callwithus/44)
exten => s,n,Wait(2)
exten => s,n,Hangup()
~


On Sun, Dec 5, 2010 at 8:34 PM, Steve Edwards  wrote:
> On Sun, 5 Dec 2010, Thomas Perron wrote:
>
>> Any reason why I don't get audio on the channel after it rings and the
>> end user picks up.
>
>> exten => s,n,Dial(SIP/callwithus/44,120,A,(demo-thanks))
>
> Re-read 'core show application dial'
>
> Where is your prompt to option 'A' ?
>
> --
> Thanks in advance,
> -
> Steve Edwards       sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
> Newline                                              Fax: +1-760-731-3000
>
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Re: [asterisk-users] no audio

[Steve Edwards]

On Sun, 5 Dec 2010, Thomas Perron wrote:

> Any reason why I don't get audio on the channel after it rings and the
> end user picks up.

> exten => s,n,Dial(SIP/callwithus/44,120,A,(demo-thanks))

Re-read 'core show application dial'

Where is your prompt to option 'A' ?

-- 
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-
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Newline  Fax: +1-760-731-3000

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[asterisk-users] no audio

[Thomas Perron]

Any reason why I don't get audio on the channel after it rings and the
end user picks up.
Here are my files.


CONSOLE=Console/dsp ; Console interface for demo
OUTBOUNDTRUNK=SIP/callwithus

[default]
include => stdexten
exten => s,1,Answer()
exten => s,n,Wait(1)
exten => s,n,Dial(SIP/callwithus/44,120,A,(demo-thanks))
exten => s,n,Wait(2)
exten => s,n,Hangup()




my sip.conf file

[general]
context=default
allowoverlap=no
bindport=5060
port=5060
bindaddr=0.0.0.0
canreinvite=no ;if your asterisk box is behind a NAT ro

;register => xxx:y...@carrier.callwithus.com
register => xxx:y...@sip.callwithus.com

[callwithus]
type=friend
host=sip.callwithus.com
username=xxx
secret=yyy
qualify=no
insecure=invite

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[asterisk-users] No audio with gtalk client behind http proxy

[Gustavo Garcia Bernardo]

Hi all,

I'm trying to establish jingle call in this network scenario:

Asterisk -> NAT -> Internet -> HTTP_PROXY -> GTalk  client

The call is received and answered in gtalk but there is no audio in the call.   
I suppose it could be related to the support for relay candidates in asterisk 
jingle implementation.Anybody else has faced this problem?

Notes: Asterisk 1.6.2. It works fine with natted gtalk clients not being behind 
proxies.

G.
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Re: [asterisk-users] No audio on call forward after upgrade from Asterisk 1.4 to 1.6

[Alex Ferrara]

Hi Paul,

No cigar unfortunately. I also tried encoding the message as gsm, ulaw and alaw 
with no success.

The ISDN interface is alaw and the SIP phones I was testing with are definately 
alaw.

Not sure what to do from here. I might just need to bypass the issue using some 
alternate way to put the message in front of the inbound dialplan logic on some 
condition.

aF

On 01/09/2010, at 8:06 AM, Paul Belanger wrote:

> On Tue, Aug 31, 2010 at 5:50 PM, Alex Ferrara  wrote:
>> Hi Paul,
>> 
>> I tried adding Progress() to no avail. I still get no audio and below is 
>> what comes up in the console.
>> 
> Try moving Progress() before the Dial().  If you Answer() the channel,
> do you have the same problem?
> 
> -- 
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
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> 
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Re: [asterisk-users] No audio on call forward after upgrade fromAsterisk 1.4 to 1.6

[Alex Ferrara]

Hi Danny,

I don't think this is the issue as I get the same problem when I divert one of 
my SIP handsets to that extension, and dial internally. The connection happens 
instantly. I can see the file playing on the asterisk console whilst I am 
getting dead air.

aF

On 01/09/2010, at 7:54 AM, Danny Nicholas wrote:

> You're probably not going to buy this, but if custom/ceh-meetingmsg is less
> than 7 seconds long, it could be playing before the connection is
> established.
> 
> 
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Re: [asterisk-users] No audio on call forward after upgrade from Asterisk 1.4 to 1.6

[Paul Belanger]

On Tue, Aug 31, 2010 at 5:50 PM, Alex Ferrara  wrote:
> Hi Paul,
>
> I tried adding Progress() to no avail. I still get no audio and below is what 
> comes up in the console.
>
Try moving Progress() before the Dial().  If you Answer() the channel,
do you have the same problem?

-- 
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Re: [asterisk-users] No audio on call forward after upgrade fromAsterisk 1.4 to 1.6

[Danny Nicholas]

You're probably not going to buy this, but if custom/ceh-meetingmsg is less
than 7 seconds long, it could be playing before the connection is
established.


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Re: [asterisk-users] No audio on call forward after upgrade from Asterisk 1.4 to 1.6

[Alex Ferrara]

Hi Paul,

I tried adding Progress() to no avail. I still get no audio and below is what 
comes up in the console.

 -- Accepting call from '403xx' to '0812' on channel 0/10, span 1
-- Executing [0...@isdn-incoming:1] Dial("DAHDI/10-1", "SIP/812,60") in 
new stack
  == Using SIP RTP CoS mark 5
-- Called 812
-- Got SIP response 302 "Moved Temporarily" back from 192.168.1.148
-- Now forwarding DAHDI/10-1 to 'Local/8...@smallanimals' (thanks to 
SIP/812-0016)
-- Executing [...@smallanimals:1] 
Progress("Local/8...@smallanimals-21bd;2", "") in new stack
-- Executing [...@smallanimals:2] 
Playback("Local/8...@smallanimals-21bd;2", "custom/ceh-meetingmsg") in new stack
--  Playing 'custom/ceh-meetingmsg.gsm' 
(language 'en')
-- Channel 0/10, span 1 got hangup request, cause 16
  == Spawn extension (isdn-incoming, 0812, 1) exited non-zero on 
'DAHDI/10-1'
-- Hungup 'DAHDI/10-1'
  == Spawn extension (smallanimals, 849, 2) exited non-zero on 
'Local/8...@smallanimals-21bd;2'

The notion brought up earlier of a codec mismatch and Asterisk not transcoding 
feels like the right answer, but I won't know until I get on site.

Thanks for the reply.

aF

On 31/08/2010, at 10:47 PM, Paul Belanger wrote:

> On Tue, Aug 31, 2010 at 4:04 AM, Alex Ferrara  wrote:
>> exten => 849,1,Playback(custom/ceh-meetingmsg)
>> exten => 849,n,Hangup
>> 
> exten => 849,1,Progress()
> exten => 849,n,Playback(custom/ceh-meetingmsg)
> exten => 849,n,Hangup
> 
> -- 
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> Polybeacon | Consultant
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> 
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Re: [asterisk-users] No audio on call forward after upgrade from Asterisk 1.4 to 1.6

[Paul Belanger]

On Tue, Aug 31, 2010 at 4:04 AM, Alex Ferrara  wrote:
> exten => 849,1,Playback(custom/ceh-meetingmsg)
> exten => 849,n,Hangup
>
exten => 849,1,Progress()
exten => 849,n,Playback(custom/ceh-meetingmsg)
exten => 849,n,Hangup

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Re: [asterisk-users] No audio on call forward after upgrade from Asterisk 1.4 to 1.6

[Ondrej Škopek]

Hi Alex,


I'm new to this list, but I had this problem too, and I solved it looking at
the codecs the sip handsets use, and then I converted the voice prompts to
that codec just like Philipp said..

Ondrej

On Tue, Aug 31, 2010 at 10:04 AM, Alex Ferrara wrote:

> Hi everyone,
>
> This is my first post to the list, although I am a long term user of
> Asterisk. I have recently found a problem that I just can't seem to solve.
>
> I have a client that has an Ubuntu x64 based Asterisk server with and ISDN
> Dahdi interface and about 25 SIP handsets. Everything was working fine in
> Asterisk 1.4 and now after migrating the config to Asterisk 1.6.2.5 I have
> one single issue that I can't explain.
>
> I have an extension that if you call it, it will play a sound file and
> hangup. Pretty simple stuff. Below is the extensions.conf entry for this
> extension.
>
> exten => 849,1,Playback(custom/ceh-meetingmsg)
> exten => 849,n,Hangup
>
> The following happens if I dial it from a SIP handset
>
>  == Using SIP RTP CoS mark 5
>-- Executing [...@smallanimals:1] Playback("SIP/812-0074",
> "custom/ceh-meetingmsg") in new stack
>--  Playing 'custom/ceh-meetingmsg.gsm' (language
> 'en')
>-- Executing [...@smallanimals:2] Hangup("SIP/812-0074", "") in new
> stack
>  == Spawn extension (smallanimals, 849, 2) exited non-zero on
> 'SIP/812-0074'
>
> The scenario is during the day, if my client has a staff meeting, they
> simply turn on call forwarding on the reception phone to this extension. In
> the past, the audio would start as soon as the caller dials in.
>
> After upgrading to Asterisk 1.6, we simply get no audio until the dialplan
> finishes. On the Asterisk console, I can see that the sound file is indeed
> playing, but we can't hear it. This happens if I am dialing the from a SIP
> extension on the phone system, or if I dial in from the public phone system.
>
>  == Using SIP RTP CoS mark 5
>-- Executing [...@smallanimals:1] Dial("SIP/811-0046",
> "SIP/812,60") in new stack
>  == Using SIP RTP CoS mark 5
>-- Called 812
>-- Got SIP response 302 "Moved Temporarily" back from 192.168.1.148
>-- Now forwarding SIP/811-0046 to 'Local/8...@smallanimals' (thanks
> to SIP/812-0047)
>-- Executing [...@smallanimals:1] 
> Playback("Local/8...@smallanimals-b5dd;2",
> "custom/ceh-meetingmsg") in new stack
>--  Playing 'custom/ceh-meetingmsg.gsm'
> (language 'en')
>
> I have tried so many things that I have lost count, and I humbly ask the
> collective intelligence of the Asterisk community for assistance.
>
> Many thanks
>
> aF
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Re: [asterisk-users] No audio on call forward after upgrade from Asterisk 1.4 to 1.6

[Philipp von Klitzing]

Hi!

> After upgrading to Asterisk 1.6, we simply get no audio until the dialplan
> finishes. On the Asterisk console, I can see that the sound file is indeed
> playing, but we can't hear it. [...]
> 
> I have tried so many things that I have lost count, and I humbly ask the
> collective intelligence of the Asterisk community for assistance.

For a start: 

* upgarde to the current release of 1.6.2.x
* does that message play when you call it without a forward (302) on your 
admin phone?
* convert the .gsm prompt to a .wav or .alaw or .ulaw prompt and see if 
that improves matters
* do a "RTP debug" to see if there is any RTP being sent at all
* consider ChanSpy for listening in (although I doubt that'll help you)

Philipp


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[asterisk-users] No audio on call forward after upgrade from Asterisk 1.4 to 1.6

[Alex Ferrara]

Hi everyone,

This is my first post to the list, although I am a long term user of Asterisk. 
I have recently found a problem that I just can't seem to solve.

I have a client that has an Ubuntu x64 based Asterisk server with and ISDN 
Dahdi interface and about 25 SIP handsets. Everything was working fine in 
Asterisk 1.4 and now after migrating the config to Asterisk 1.6.2.5 I have one 
single issue that I can't explain.

I have an extension that if you call it, it will play a sound file and hangup. 
Pretty simple stuff. Below is the extensions.conf entry for this extension.

exten => 849,1,Playback(custom/ceh-meetingmsg)
exten => 849,n,Hangup

The following happens if I dial it from a SIP handset

  == Using SIP RTP CoS mark 5
-- Executing [...@smallanimals:1] Playback("SIP/812-0074", 
"custom/ceh-meetingmsg") in new stack
--  Playing 'custom/ceh-meetingmsg.gsm' (language 'en')
-- Executing [...@smallanimals:2] Hangup("SIP/812-0074", "") in new 
stack
  == Spawn extension (smallanimals, 849, 2) exited non-zero on 
'SIP/812-0074'

The scenario is during the day, if my client has a staff meeting, they simply 
turn on call forwarding on the reception phone to this extension. In the past, 
the audio would start as soon as the caller dials in.

After upgrading to Asterisk 1.6, we simply get no audio until the dialplan 
finishes. On the Asterisk console, I can see that the sound file is indeed 
playing, but we can't hear it. This happens if I am dialing the from a SIP 
extension on the phone system, or if I dial in from the public phone system.

 == Using SIP RTP CoS mark 5
-- Executing [...@smallanimals:1] Dial("SIP/811-0046", "SIP/812,60") in 
new stack
  == Using SIP RTP CoS mark 5
-- Called 812
-- Got SIP response 302 "Moved Temporarily" back from 192.168.1.148
-- Now forwarding SIP/811-0046 to 'Local/8...@smallanimals' (thanks to 
SIP/812-0047)
-- Executing [...@smallanimals:1] 
Playback("Local/8...@smallanimals-b5dd;2", "custom/ceh-meetingmsg") in new stack
--  Playing 'custom/ceh-meetingmsg.gsm' 
(language 'en')

I have tried so many things that I have lost count, and I humbly ask the 
collective intelligence of the Asterisk community for assistance.

Many thanks

aF
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[asterisk-users] No audio using xlite

[Janu Mukherjee]

Hi,
I installed asterisk server in my system running linux. I configured a user
1000 using xlite and registered with asterisk server in the same linux
system. I configured one more user 1001 in another linux machine and this
user also got registered with asterisk. But i am facing two issues here.
   1. When a call is made from 1001 to 1000 i could see an incoming call
   blinking but no audio flow is observed.
   2. When i made a call from 1000 to 1001 it is showing incoming on line 3
of
   1000. What could be the problem.
   I wrote the dial plan as follows.
   [default]
   exten=>1000,1,Dial(SIP/1000)
   exten=>1001,1,Dial(SIP/1001)
   Can anyone please help me to solve this.
   Thanks in Advance,
   Saritha.
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Re: [asterisk-users] No audio using xlite

[Randy R]

On Sun, Jul 25, 2010 at 10:20 PM, Janu Mukherjee  wrote:

> I installed asterisk server in my linux box. I configured a user 1000 using
> xlite and registered with asterisk server in the same linux box. I

Where on the network is this box?

> configured one more user 1001 in other box and this user also got registered
> with asterisk. But i am facing two issues here.

Where on the network is this "other" box?

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[asterisk-users] No audio using xlite

[Janu Mukherjee]

Hi,

I installed asterisk server in my linux box. I configured a user 1000 using
xlite and registered with asterisk server in the same linux box. I
configured one more user 1001 in other box and this user also got registered
with asterisk. But i am facing two issues here.

1. When a call is made from 1001 to 1000 i could see an incoming call
blinking but no audio flow is observed.
2. When i made a call from 1000 to 1001 it is showing incoming on line 3 of
1000. What could be the problem.

I wrote the dial plan as follows.

[default]
exten=>1000,1,Dial(SIP/1000)
exten=>1001,1,Dial(SIP/1001)

Can anyone please help me to solve this.

Thanks in Advance,
Saritha.
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[asterisk-users] No audio when calling via PSTN, before remote answers (with polarity reversal)

[Luar Roji]

Hi!

I want to get audio from the PSTN before the call is answered so I don't miss
when the called phone is busy or if there is some error (like the phone is
unavailable or is wrong, etc) and hear the ringing from my telco.

I have polarity reversal in my telco for incoming and outgoing calls. 

If I set answeronpolarityswitch=yes then I get no audio until the call is
answered. If I set it to "no" it works fine sometimes, but other times when 
the call gets answered, asterisk detects the polarity reversal as a 
hangup and hangs up the call.

I need to have hanguponpolarityswitch set to yes for detecting hang ups in
incoming calls. (it was a nightmare before this)

Any ideas?

I had this working in previous versions of asterisk but didn't find what
was the change that caused this behaviour, and I need to use a recent version
of asterisk for some changes in dtmf caller id detection that aren't in my
distro yet (debian).

Thanks in advance.

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Re: [asterisk-users] No Audio on pstn call

[Siti Zalifah Md Yatim]

Hi IRFAN,

Thanks for that, actually, I think my FXO card already struck by
lightning. I;ve changed to another card, and now work like charm.

-

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Re: [asterisk-users] No Audio on pstn call

[LATEEF, IRFAN (ATTSI)]

Try setting the debug level higher, it might give more info to debug .

Core set debug atleast 17
Core set verbose atleast 17

-Irfan

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Siti Zalifah Md 
Yatim
Sent: Thursday, March 04, 2010 1:51 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] No Audio on pstn call

additional info on the system

Linux home 2.6.30.3-SLACKWARE #1 Sun Feb 7 09:09:33 MYT 2010 i686
Intel(R) Pentium(R) 4 CPU 2.00GHz GenuineIntel GNU/Linux


Asterisk 1.6.2.5, Copyright (C) 1999 - 2009 Digium, Inc. and others.
Created by Mark Spencer 
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.2.5 currently running on home (pid = 11838)
Verbosity is at least 7



home*CLI> module show like dahdi
Module Description
 Use Count
codec_dahdiGeneric DAHDI Transcoder Codec Translato 0
app_dahdibarge.so  Barge in on DAHDI channel application0
chan_dahdi.so  DAHDI Telephony Driver   0
app_dahdiscan.so   Scan DAHDI channels application  0
app_dahdiras.soDAHDI ISDN Remote Access Server  0
res_timing_dahdi.soDAHDI Timing Interface   0

on the other hand, calls made internally are ok.



On Thu, Mar 4, 2010 at 2:43 PM, Siti Zalifah Md Yatim
 wrote:
> Hello,
>
> I'm facing problem where as whenever there are incoming call from
> pstn, there will be no audio coming in. User at the other end also
> could not hear my voice. This happens few days back. Im using asterisk
> 1.6.1.2 with dahdi tool 2.2.0.
>
> I thought it was time to upgrade, so upgraded to dahdi 2.2.1 and
> asterisk 1.6.2.5. However, it does not help at all.
>
> My current config as follows :-
>
> X100P clone card
>
> /etc/dahdi/system.conf
> # Span 1: WCFXO/0 "Generic Clone Board 1" (MASTER)
> fxsks=1
> echocanceller=mg2,1
>
>
> /etc/asterisk/dahdi-channels.conf
> ; Span 1: WCFXO/0 "Generic Clone Board 1" (MASTER)
> ;;; line="1 WCFXO/0/0 FXSKS  (SWEC: MG2)"
> signalling=fxs_ls
> callerid=asreceived
> group=0
> context=from-pstn
> channel => 1
> callerid=
> group=
> context=default
>
>
> /etc/asterisk/chan_dahdi.conf
>
> [trunkgroups]
>
>
>
>
> [channels]
> language = my
> ;
> usecallerid = yes
> callwaiting = yes
> usecallingpres = yes
> callwaitingcallerid = yes
> threewaycalling = yes
> transfer = yes
> canpark = yes
> cancallforward = yes
> callreturn = yes
> mailbox = 5000
> echocancel = yes
> echocancelwhenbridged = yes
> rxgain = 2.0
> txgain = 3.0
> group = 1
> callgroup = 1
> pickupgroup = 1
> faxdetect = both
> signalling = fxs_ls
> callerid = asreceived
> group = 0
> channel = 1
> callerid =
> group =
> context = default
> #include "dahdi-channels.conf"
>
>
> my call plan will execute voicemail when there;s incoming call from
> pstn. result as shwon here
>
>
> -- Executing [...@from-pstn:1] Set("DAHDI/1-1", "CallTime=20100304
> 13:45:30") in new stack
> -- Executing [...@from-pstn:2] Set("DAHDI/1-1", "CallerIDString=""
> <01935x>") in new stack
> -- Executing [...@from-pstn:3] System("DAHDI/1-1", "/bin/echo "20100304
> 13:45:30 01935x [] - to pstn" >> /var/log/asterisk/call_log") in
> new stack
> -- Executing [...@from-pstn:4] Answer("DAHDI/1-1", "") in new stack
> -- Executing [...@from-pstn:5] VoiceMail("DAHDI/1-1", "5000,u") in new stack
> -- Stopped music on hold on DAHDI/1-1
> -- Playing 'vm-theperson.gsm' (language 'my')
> -- Playing 'digits/5.gsm' (language 'my')
> -- Playing 'digits/0.gsm' (language 'my')
> -- Playing 'digits/0.gsm' (language 'my')
> -- Playing 'digits/0.gsm' (language 'my')
> -- Playing 'vm-isunavail.gsm' (language 'my')
> -- Playing 'vm-intro.gsm' (language 'my')
> -- Playing 'beep.gsm' (language 'my')
> -- Recording the mes

Re: [asterisk-users] No Audio on pstn call

[Siti Zalifah Md Yatim]

additional info on the system

Linux home 2.6.30.3-SLACKWARE #1 Sun Feb 7 09:09:33 MYT 2010 i686
Intel(R) Pentium(R) 4 CPU 2.00GHz GenuineIntel GNU/Linux


Asterisk 1.6.2.5, Copyright (C) 1999 - 2009 Digium, Inc. and others.
Created by Mark Spencer 
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.2.5 currently running on home (pid = 11838)
Verbosity is at least 7



home*CLI> module show like dahdi
Module Description
 Use Count
codec_dahdiGeneric DAHDI Transcoder Codec Translato 0
app_dahdibarge.so  Barge in on DAHDI channel application0
chan_dahdi.so  DAHDI Telephony Driver   0
app_dahdiscan.so   Scan DAHDI channels application  0
app_dahdiras.soDAHDI ISDN Remote Access Server  0
res_timing_dahdi.soDAHDI Timing Interface   0

on the other hand, calls made internally are ok.



On Thu, Mar 4, 2010 at 2:43 PM, Siti Zalifah Md Yatim
 wrote:
> Hello,
>
> I'm facing problem where as whenever there are incoming call from
> pstn, there will be no audio coming in. User at the other end also
> could not hear my voice. This happens few days back. Im using asterisk
> 1.6.1.2 with dahdi tool 2.2.0.
>
> I thought it was time to upgrade, so upgraded to dahdi 2.2.1 and
> asterisk 1.6.2.5. However, it does not help at all.
>
> My current config as follows :-
>
> X100P clone card
>
> /etc/dahdi/system.conf
> # Span 1: WCFXO/0 "Generic Clone Board 1" (MASTER)
> fxsks=1
> echocanceller=mg2,1
>
>
> /etc/asterisk/dahdi-channels.conf
> ; Span 1: WCFXO/0 "Generic Clone Board 1" (MASTER)
> ;;; line="1 WCFXO/0/0 FXSKS  (SWEC: MG2)"
> signalling=fxs_ls
> callerid=asreceived
> group=0
> context=from-pstn
> channel => 1
> callerid=
> group=
> context=default
>
>
> /etc/asterisk/chan_dahdi.conf
>
> [trunkgroups]
>
>
>
>
> [channels]
> language = my
> ;
> usecallerid = yes
> callwaiting = yes
> usecallingpres = yes
> callwaitingcallerid = yes
> threewaycalling = yes
> transfer = yes
> canpark = yes
> cancallforward = yes
> callreturn = yes
> mailbox = 5000
> echocancel = yes
> echocancelwhenbridged = yes
> rxgain = 2.0
> txgain = 3.0
> group = 1
> callgroup = 1
> pickupgroup = 1
> faxdetect = both
> signalling = fxs_ls
> callerid = asreceived
> group = 0
> channel = 1
> callerid =
> group =
> context = default
> #include "dahdi-channels.conf"
>
>
> my call plan will execute voicemail when there;s incoming call from
> pstn. result as shwon here
>
>
> -- Executing [...@from-pstn:1] Set("DAHDI/1-1", "CallTime=20100304
> 13:45:30") in new stack
> -- Executing [...@from-pstn:2] Set("DAHDI/1-1", "CallerIDString=""
> <01935x>") in new stack
> -- Executing [...@from-pstn:3] System("DAHDI/1-1", "/bin/echo "20100304
> 13:45:30 01935x [] - to pstn" >> /var/log/asterisk/call_log") in
> new stack
> -- Executing [...@from-pstn:4] Answer("DAHDI/1-1", "") in new stack
> -- Executing [...@from-pstn:5] VoiceMail("DAHDI/1-1", "5000,u") in new stack
> -- Stopped music on hold on DAHDI/1-1
> -- Playing 'vm-theperson.gsm' (language 'my')
> -- Playing 'digits/5.gsm' (language 'my')
> -- Playing 'digits/0.gsm' (language 'my')
> -- Playing 'digits/0.gsm' (language 'my')
> -- Playing 'digits/0.gsm' (language 'my')
> -- Playing 'vm-isunavail.gsm' (language 'my')
> -- Playing 'vm-intro.gsm' (language 'my')
> -- Playing 'beep.gsm' (language 'my')
> -- Recording the message
> -- x=0, open writing:
> /var/spool/asterisk/voicemail/default/5000/tmp/ksOvKw format: wav,
> 0x91bfb68
> -- Recording automatically stopped after a silence of 10 seconds
> -- Playing 'auth-thankyou.gsm' (language 'my')
> -- Executing [...@from-pstn:8] Hangup("DAHDI/1-1", "") in new stack
>
> how ever,
> starting from line 5 onwards, theres no audio at all.
>
> anybody can help ?
>
> thank you.
>

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[asterisk-users] No Audio on pstn call

[Siti Zalifah Md Yatim]

Hello,

I'm facing problem where as whenever there are incoming call from
pstn, there will be no audio coming in. User at the other end also
could not hear my voice. This happens few days back. Im using asterisk
1.6.1.2 with dahdi tool 2.2.0.

I thought it was time to upgrade, so upgraded to dahdi 2.2.1 and
asterisk 1.6.2.5. However, it does not help at all.

My current config as follows :-

X100P clone card

/etc/dahdi/system.conf
# Span 1: WCFXO/0 "Generic Clone Board 1" (MASTER)
fxsks=1
echocanceller=mg2,1


/etc/asterisk/dahdi-channels.conf
; Span 1: WCFXO/0 "Generic Clone Board 1" (MASTER)
;;; line="1 WCFXO/0/0 FXSKS  (SWEC: MG2)"
signalling=fxs_ls
callerid=asreceived
group=0
context=from-pstn
channel => 1
callerid=
group=
context=default


/etc/asterisk/chan_dahdi.conf

[trunkgroups]




[channels]
language = my
;
usecallerid = yes
callwaiting = yes
usecallingpres = yes
callwaitingcallerid = yes
threewaycalling = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
mailbox = 5000
echocancel = yes
echocancelwhenbridged = yes
rxgain = 2.0
txgain = 3.0
group = 1
callgroup = 1
pickupgroup = 1
faxdetect = both
signalling = fxs_ls
callerid = asreceived
group = 0
channel = 1
callerid =
group =
context = default
#include "dahdi-channels.conf"


my call plan will execute voicemail when there;s incoming call from
pstn. result as shwon here


-- Executing [...@from-pstn:1] Set("DAHDI/1-1", "CallTime=20100304
13:45:30") in new stack
-- Executing [...@from-pstn:2] Set("DAHDI/1-1", "CallerIDString=""
<01935x>") in new stack
-- Executing [...@from-pstn:3] System("DAHDI/1-1", "/bin/echo "20100304
13:45:30 01935x [] - to pstn" >> /var/log/asterisk/call_log") in
new stack
-- Executing [...@from-pstn:4] Answer("DAHDI/1-1", "") in new stack
-- Executing [...@from-pstn:5] VoiceMail("DAHDI/1-1", "5000,u") in new stack
-- Stopped music on hold on DAHDI/1-1
-- Playing 'vm-theperson.gsm' (language 'my')
-- Playing 'digits/5.gsm' (language 'my')
-- Playing 'digits/0.gsm' (language 'my')
-- Playing 'digits/0.gsm' (language 'my')
-- Playing 'digits/0.gsm' (language 'my')
-- Playing 'vm-isunavail.gsm' (language 'my')
-- Playing 'vm-intro.gsm' (language 'my')
-- Playing 'beep.gsm' (language 'my')
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/5000/tmp/ksOvKw format: wav,
0x91bfb68
-- Recording automatically stopped after a silence of 10 seconds
-- Playing 'auth-thankyou.gsm' (language 'my')
-- Executing [...@from-pstn:8] Hangup("DAHDI/1-1", "") in new stack

how ever,
starting from line 5 onwards, theres no audio at all.

anybody can help ?

thank you.

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[asterisk-users] No audio - using g729 codec altogether

[ast guy]

Hi,
 I am facing terrible issue regarding no audio/voice on both sides. I am
using g729 codec on two machines and carrier also supports g729 codec. I can
see the RTP traffic flowing but there is no audio.
Call is going from Server 1 to Server 2. I can see the established SIP
channels on Server but still no audio

C.L.M.37 = Global Address **-Server-1*
X.Y.X.55 = LAN Address of **-Server-1*
M.G.W.23 = Media Gateway of Carrier
A.B.C.136 = Global Address **-Server-2*


**-Server-1*
codec and format used:

codec_g729.so
format_g729.so

126.475451X.Y.X.55 -> M.G.W.23RTP Payload type=ITU-T G.729,
SSRC=1224682667, Seq=171, Time=171288
126.495804 A.B.C.136 -> C.L.M.37 RTP Payload type=ITU-T G.729,
SSRC=1406269818, Seq=17753, Time=171608
126.495833X.Y.X.55 -> M.G.W.23RTP Payload type=ITU-T G.729,
SSRC=1224682667, Seq=172, Time=171448
126.515405 A.B.C.136 -> C.L.M.37 RTP Payload type=ITU-T G.729,
SSRC=1406269818, Seq=17754, Time=171768
126.515435X.Y.X.55 -> M.G.W.23RTP Payload type=ITU-T G.729,
SSRC=1224682667, Seq=173, Time=171608
126.535204 A.B.C.136 -> C.L.M.37 RTP Payload type=ITU-T G.729,
SSRC=1406269818, Seq=17755, Time=171928
126.535423X.Y.X.55 -> M.G.W.23RTP Payload type=ITU-T G.729,
SSRC=1224682667, Seq=174, Time=171768
126.555461 A.B.C.136 -> C.L.M.37 RTP Payload type=ITU-T G.729,
SSRC=1406269818, Seq=17756, Time=172088
126.79X.Y.X.55 -> M.G.W.23RTP Payload type=ITU-T G.729,
SSRC=1224682667, Seq=175, Time=171928


**-Server-2*

Codec used : codec_g729-ast12-gcc4-glibc-x86_64-pentium4.so

101.374796 A.B.C.136 -> C.L.M.37 RTP PT=ITU-T G.729, SSRC=0x53D1F97A,
Seq=17785, Time=176728
101.389644 213.166.5.134 -> A.B.C.136 RTP PT=ITU-T G.729, SSRC=0x15130586,
Seq=55472, Time=879827237
101.389665 A.B.C.136 -> C.L.M.37 RTP PT=ITU-T G.729, SSRC=0x53D1F97A,
Seq=17786, Time=176888
101.409653 213.166.5.134 -> A.B.C.136 RTP PT=ITU-T G.729, SSRC=0x15130586,
Seq=55473, Time=879827397
101.409674 A.B.C.136 -> C.L.M.37 RTP PT=ITU-T G.729, SSRC=0x53D1F97A,
Seq=17787, Time=177048
101.429709 213.166.5.134 -> A.B.C.136 RTP PT=ITU-T G.729, SSRC=0x15130586,
Seq=55474, Time=879827557
101.429723 A.B.C.136 -> C.L.M.37 RTP PT=ITU-T G.729, SSRC=0x53D1F97A,
Seq=17788, Time=177208
101.454956 213.166.5.134 -> A.B.C.136 RTP PT=ITU-T G.729, SSRC=0x15130586,
Seq=55475, Time=879827717


Any one has any idea why it is behaving so.


/ag
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Re: [asterisk-users] No audio on remote SIP calls

[Benny Amorsen]

Jonathan Moore  writes:

> The idea of RTP being to blame would make sense though.  I can
> still transfer and such, and watching the console, I see when I press
> various keys on the phone, so it seems that the SIP traffic is working
> out fine.  (I do understand that right?  SIP == control RTP == voice
> in a very generic sense?)

> I plan to take a packet trace in the morning on the asterisk server and
> see what is going on at that level.  Hints as to what I should be looking
> for?

Start by looking at pure SIP traffic by doing -s0 -v and filtering on port
5060. Notice the media streams being negotiated, and look at the IP
addresses and ports.

If that doesn't help, remove the port 5060 filter and look again at the
raw traffic -- but that can be a lot of traffic.

My guess: You have STUN enabled on the phones.


/Benny


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Re: [asterisk-users] No audio on remote SIP calls

[Ishfaq Malik]

Hi

The only time I've had issues that seem a bit like yours it was down to 
the order of codecs in the handset settings. Make sure they match the 
order dictated on the server.

Ish

Jonathan Moore wrote:
> On Thu, Aug 6, 2009 at 5:17 PM, SŽébastien
> Cramatte wrote:
>   
>> Hi,
>>
>> This sounds udp RTP problem.
>> Might be you have some firewall rules that block this kind of traffic ?
>> As soon I remember, Asterisk by default use random port between 1
>> and 2 for rtp traffic (you can adjust this in rtp.conf).
>> 
>
> In theory, there should be no firewalls between my asterisk server
> and the remote phones.  I've opened a ticket with ATT with that
> exact question, as well as a question of rather any NATing is going
> on, though, I doubt this is the case, and this is the first time this type
> of problem has happened in over 4 years.
>
> The idea of RTP being to blame would make sense though.  I can
> still transfer and such, and watching the console, I see when I press
> various keys on the phone, so it seems that the SIP traffic is working
> out fine.  (I do understand that right?  SIP == control RTP == voice
> in a very generic sense?)
>
> I plan to take a packet trace in the morning on the asterisk server and
> see what is going on at that level.  Hints as to what I should be looking
> for?
>
> -jonathan
>
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>
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-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

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Re: [asterisk-users] No audio on remote SIP calls

[Jonathan Moore]

On Thu, Aug 6, 2009 at 5:17 PM, SŽébastien
Cramatte wrote:
> Hi,
>
> This sounds udp RTP problem.
> Might be you have some firewall rules that block this kind of traffic ?
> As soon I remember, Asterisk by default use random port between 1
> and 2 for rtp traffic (you can adjust this in rtp.conf).

In theory, there should be no firewalls between my asterisk server
and the remote phones.  I've opened a ticket with ATT with that
exact question, as well as a question of rather any NATing is going
on, though, I doubt this is the case, and this is the first time this type
of problem has happened in over 4 years.

The idea of RTP being to blame would make sense though.  I can
still transfer and such, and watching the console, I see when I press
various keys on the phone, so it seems that the SIP traffic is working
out fine.  (I do understand that right?  SIP == control RTP == voice
in a very generic sense?)

I plan to take a packet trace in the morning on the asterisk server and
see what is going on at that level.  Hints as to what I should be looking
for?

-jonathan

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Re: [asterisk-users] No audio on remote SIP calls

[SŽébastien Cramatte]

Hi,

This sounds udp RTP problem.
Might be you have some firewall rules that block this kind of traffic ?
As soon I remember, Asterisk by default use random port between 1 
and 2 for rtp traffic (you can adjust this in rtp.conf).

- Sebastien


Jonathan Moore escribió:
> Hi everyone.
>
> We have an asterisk server in our main office and phones at each
> remote site.  The remote offices are connected via a MPLS which, to my
> knowledge has no natting going on.
>
> The problem I have is that any call from a remote phone to a remote
> phone (even on the same remote lan) results in no audio.  If I make a
> call from the same LAN the asterisk server is on, to one of these
> remote sites, I get perfect two way audio.  If I play a call from one
> phone to another at a remote site, there is no audio, however, I do
> hear messages (such as voicemail, things from Playback(), etc) that
> originate on the asterisk server.
>
> I've tried adjusting canreinvite= in sip.conf in hopes in might have
> some effect, but so far nothing.
>
> Suggestions on where else to look, or what the problem might be?
>
> Which configs would be useful in troubleshooting?
>
> Thanks.
>
> -jonathan
>
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[asterisk-users] No audio on remote SIP calls

[Jonathan Moore]

Hi everyone.

We have an asterisk server in our main office and phones at each
remote site.  The remote offices are connected via a MPLS which, to my
knowledge has no natting going on.

The problem I have is that any call from a remote phone to a remote
phone (even on the same remote lan) results in no audio.  If I make a
call from the same LAN the asterisk server is on, to one of these
remote sites, I get perfect two way audio.  If I play a call from one
phone to another at a remote site, there is no audio, however, I do
hear messages (such as voicemail, things from Playback(), etc) that
originate on the asterisk server.

I've tried adjusting canreinvite= in sip.conf in hopes in might have
some effect, but so far nothing.

Suggestions on where else to look, or what the problem might be?

Which configs would be useful in troubleshooting?

Thanks.

-jonathan

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[asterisk-users] No Audio PlayBack Asterisk 1.6 Dahdi 2.1.0.3

[Daviramos Roussenq Fortunato]

Hi List.


I'm having problems with Asterisk 1.6 + DAHDI 2.1.0.3

PlayBack does not ring, is still in command, and not later in the following
context.

Disabling the dahdi operates normally.

I'm using dahdi_dummy.
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Re: [asterisk-users] No Audio

[michel freiha]

Dear Sir,

I used several other Softphones like Skype and they are facing the same
problem...It seems that the issue is global du to an undersea cable cut
Regards

On Mon, Dec 22, 2008 at 9:07 PM, michel freiha  wrote:

> Hi all,
> Sometimes when making a PC to PSTN call through asterisk, I got no audio in
> both sides...tracing by wireshark, I can find that RTP packets are hitting
> my PC but no audio...Can someone guess what could be that issue?
>
> Maybe it's a latency issue?
>
> Regards
>
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[asterisk-users] No Audio

[michel freiha]

Hi all,
Sometimes when making a PC to PSTN call through asterisk, I got no audio in
both sides...tracing by wireshark, I can find that RTP packets are hitting
my PC but no audio...Can someone guess what could be that issue?

Maybe it's a latency issue?

Regards
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[asterisk-users] No audio after transferring to voicemail

[Jeremy Phillips]

Hello All,

I'm having an issue where asterisk doesn't hear any audio after transferring to 
voicemail. Here is the dial plan and console output.

DIAL PLAN
[voicepulse-in]
exten => _14259491337,1,NoOp(Incoming call from VoicePulse)
exten => _14259491337,2,Ringing
exten => _14259491337,3,Wait(1)
exten => _14259491337,4,Dial(SIP/1337,20)
exten => _14259491337,5,VoiceMail(1337)
exten => _14259491337,6,Wait(1)
exten => _14259491337,7,HangUp

CONSOLE OUTPUT
-- Executing [EMAIL PROTECTED]:4] Dial("SIP/-081d2800", "SIP/1337|20") 
in new stack
-- Called 1337
-- SIP/1337-081cfce0 is ringing
-- Nobody picked up in 2 ms
-- Executing [EMAIL PROTECTED]:5] VoiceMail("SIP/-081d2800", 
"1337") in new stack
-- -081d2800> Playing 'vm-intro' (language 'en')
-- -081d2800> Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing:  
/var/spool/asterisk/voicemail/default/1337/tmp/KzD4A1 format: wav, 0x8184358
[Oct 31 08:21:02] WARNING[22354]: app.c:602 __ast_play_and_record: No audio 
available on SIP/-081d2800??
-- User hung up
[Oct 31 08:21:02] NOTICE[22354]: pbx.c:1631 
pbx_substitute_variables_helper_full: Error in extension logic (missing '}')
  == Spawn extension (voicepulse-in, 14259491337, 5) exited non-zero on 
'SIP/-081d2800'

Any help would be greatly appreciated!

Thanks,

Jeremy Phillips
M: 540.322.7980 | T: 425.949.1337 | B: http://jeremyphillips.org

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[asterisk-users] no audio, firewall problem?

[tic tac]

Hello,

I am runing asterisk on a embedded linux and am having some RTP audio issues at 
the beginning of the call: the comfort noise packet seems to be opening the 
pinhole in the firewall though I don't understand why it is not already opened. 
Then audio is then transferred correctly between caller and callee through the 
asterisk bridge.

The SIP INVITE is received on a WAN interface and then I dial out to another 
SIP channel through the same interface. CLI output with RTP debug shows that 
Packet2Packet is only started and RTP is only sent by asterisk after the first 
rtpkeepalive timeout.

If I sniff at a mirroring port in the network I can see the first RTP packet 
going from my caller to the asterisk server yet it seems that it is never 
received (or it never reaches) asterisk (it is a direct route).

All firewall rules on the asterisk box are setup for the range of ports defined 
by rtp.conf (10k-11k in mycase); that is consistent with the SDP signaling 
generated by asterisk for the INVITE OUT and for the 200 OK back to the caller 
in the media description attribute.
Watching iptables live activation does not show any RTP packet blocked at the 
beginning of the call.

netstat shows:

netstat -an | grep udp | grep 10
netstat: no support for 'AF INET6 (tcp)' on this system
netstat: no support for 'AF INET6 (udp)' on this system
netstat: no support for 'AF INET6 (raw)' on this system
udp0  0 216.54.141.148:105540.0.0.0:*
udp0  0 216.54.141.148:105550.0.0.0:*
udp0  0 216.54.141.148:101020.0.0.0:*
udp0  0 216.54.141.148:101030.0.0.0:*

as I am using bindaddr=0.0.0.0 in the sip.conf.

I have multiple NICs on that box, could it be a problem or ...?

Thanks for any suggestion,

Sebastien.
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[asterisk-users] No Audio on Meetme

[Nhadie Ramos]

Hi All,

What could be the cause why there is no audio coming form the participants.

ztdummy is loaded, ZTDUMMY/1 (source: HRtimer) 1.

I can hear "Please enter your PIN", "User blah blah has enttered"...etc etc

But when the particpants talk, we hear nothing. What are the possible mistakes 
i did on these?

TIA
Nhadie



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Re: [asterisk-users] No audio on Sangoma A104.

[Alex Balashov]

Matt Florell wrote:

> Are you using 64bit Linux?

No.

> Do you have more than 4GB of RAM?

Indeed not.

> Have you contacted Sangoma support?

No.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] No audio on Sangoma A104.

[Matt Florell]

Are you using 64bit Linux?

Do you have more than 4GB of RAM?

Have you contacted Sangoma support?

MATT---

On 3/23/08, Alex Balashov <[EMAIL PROTECTED]> wrote:
> Alex Balashov wrote:
>
>  > I'm running kernel 2.6.19 (tried 2.6.24.3 but had to downgrade as
>  > wanpipe stuff would not compile), zaptel 1.4.9.2, and wanpipe 3.3.2.
>
>
> And Asterisk 1.4.18.1.
>
>
>
>  --
>  Alex Balashov
>  Evariste Systems
>  Web: http://www.evaristesys.com/
>  Tel: (+1) (678) 954-0670
>  Direct : (+1) (678) 954-0671
>  Mobile : (+1) (706) 338-8599
>
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Re: [asterisk-users] No audio on Sangoma A104.

[Alex Balashov]

Alex Balashov wrote:

> I'm running kernel 2.6.19 (tried 2.6.24.3 but had to downgrade as 
> wanpipe stuff would not compile), zaptel 1.4.9.2, and wanpipe 3.3.2.

And Asterisk 1.4.18.1.


-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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[asterisk-users] No audio on Sangoma A104.

[Alex Balashov]

Hi all,

I am having a very strange problem.  I am terminating a PRI (5ESS switch 
type, national plan, 23B+1D (24)) into a Sangoma A104 and am not able to 
produce any audio heard on the PSTN end of the call.

Not sure what's wrong - the card worked before under a Trixbox setup.

I'm running kernel 2.6.19 (tried 2.6.24.3 but had to downgrade as 
wanpipe stuff would not compile), zaptel 1.4.9.2, and wanpipe 3.3.2.

If anyone has any ideas, please let me know!  There is no audio, but a 
very faint stutter of echo (not audio) on the far end in correspondence 
with playback of various audio files, etc.

I've tried messing with the T1 line build-out settings and txgain out of 
the belief that perhaps the way the circuit is broken out is not 
particularly favourable (it runs between floors of a building and is 
wired from Cat-5e - don't ask, I didn't do it), but there are no errors 
on the T1 level and we've got several other PRIs designed the same way 
running into Cisco AS5300s without a problem.

Thanks!

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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