[asterisk-dev] No audio available on SIP/172.16.129.13:5060-00000001??
shalu dhamija
shalu.dhamija at rancoretech.com
Wed Jan 11 06:27:38 CST 2012
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:
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
Regards,
Shalu
Date: Wed, 11 Jan 2012 10:59:33 +0530
From: virendra bhati <virbhati at gmail.com>
Subject: Re: [asterisk-users] No audio available on
SIP/172.16.129.13:5060-00000001??
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<CANNhuhdoQvvOvvYiB7s0Pnj+OR_Xy94D0yL8fE71ekA+f4DA+w at mail.gmail.com>
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.dhamija at 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/172.16.129.13:5060-00000001??
>
>
>
> I have set rtpstart=6000 and rtpend=20000 in rtp.conf.
>
>
>
>
>
> Can someone please let me know how to avoid these kind of warnings.
>
>
>
> Thanks.
>
>
>
> Shalu
>
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