shalu dhamija
2012-Jan-11 04:35 UTC
[asterisk-users] No audio available on SIP/172.16.129.13:5060-00000001??
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 Thanks and Regards, Shalu Dhamija Rancore Technologies(P) Ltd. Gurgaon Ph : 0124-4200691 +91-9910995356(M) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120111/47ae0a24/attachment.htm>
virendra bhati
2012-Jan-11 05:29 UTC
[asterisk-users] No audio available on SIP/172.16.129.13:5060-00000001??
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 > > > > > > > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120111/731f7972/attachment.htm>
shalu dhamija
2012-Jan-11 12:27 UTC
[asterisk-users] No audio available on SIP/172.16.129.13:5060-00000001??
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>? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120111/da75a634/attachment.htm>