Hi ,
Thank you for your help Horaci I ‘ve changed the xml but it still does not
work :/
Do you have another idea why it doesn’t work ?
Thank You
Tugce
From: Horaci Macias [mailto:hmac...@avaya.com]
Sent: 22 Ağustos 2014 Cuma 13:07
To: sipp-users@lists.sourceforge.net
Subject: Re: [Sipp-users] DTMF Tone cannot be recognised by the receiver
The XML is using 110 for the telephone-event:
m=audio [auto_media_port] RTP/AVP 0 8 18 110
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:110 telephone-event/8000
but the pcaps use 101 for the rtp events; I'd suggest first trying again using
the same number in both (just search/replace 110 to 101 in the XML is probably
the easiest).
hope this helps,
H
On 20/08/14 13:55, Aksen Elcin Tugce KILINC wrote:
Hi all,
I have some trouble to send dtmf tones. .The codes are the same as in the
source page. The sender (originator ) shows that I sent the packets but the
receiver (term) does not care them. While I send the g711a.pcap ; I can hear
the tone on the term side. But when I try to send any digit or # I couldn’t
succeed .My xml file is attached as txt to the mail.
I searched on the web but I couldn’t find any solution.
Do you have any idea about it ?
Also my version is ;
sipp -v
SIPp v3.4.1-TLS-SCTP-PCAP-RTPSTREAM built Jun 17 2014, 05:26:44.
I’ll be appreciated If you help me,I really need it and I stuck here atm.
Thank you all !!
Tugce Kilinc
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Ayrıca hukuken de gizli olabilir. Hiçbir şekilde üçüncü kişilere açıklanamaz ve
yayınlanamaz. Eğer mesajın gönderildiği alıcı değilseniz bu elektronik postanın
içeriğini açıklamanız, kopyalamanız, yönlendirmeniz ve kullanmanız kesinlikle
yasaktır ve bu elektronik postayı ve eklerini derhal silmeniz gerekmektedir.
NETAŞ TELEKOMÜNİKASYON A.Ş. bu mesajın içerdiği bilgilerin doğruluğu veya
eksiksiz olduğu konusunda herhangi bir garanti vermemektedir. Bu nedenle bu
bilgilerin ne şekilde olursa olsun içeriğinden, iletilmesinden, alınmasından,
saklanmasından ve kullanılmasından sorumlu değildir. Bu mesajdaki görüşler
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the exclusive use of the individual or entity to whom it is addressed. It may
also be legally confidential. Any disclosure, distribution or other
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are not the intended recipient you are hereby notified that any dissemination,
forwarding, copying or use of any of the information is strictly prohibited,
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<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uac' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[field3]@[field0] SIP/2.0
Via: SIP/2.0/UDP [local_ip]:[local_port];branch=[branch];rport
From: "[field1]" <sip:[field1]@[field2]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[field3]@[field2]>
Contact: sip:[field1]@[local_ip]:[local_port];+sip.instance="<urn:uuid:34ac93b9-2b05-49dc-89c7-9bbfd50cec91>";reg-id=1;expires=86200;description="Login"
Call-ID: [call_id]
CSeq: [cseq] INVITE
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept-Language: tr-tr
User-Agent: A2PC 10.3.1234
Accept-Encoding: nt-im-2.0
x-nt-GUID: 0034ac93b902b05049dc089c709bbfd50cec91
x-nt-location: 4243
Min-SE: 90
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="180" >
</recv>
<!-- [auto_media_port] By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rrs="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by [remote_ip]:[remote_port] -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[field3]@[field0] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch];rport
From: "[field1]" <sip:[field1]@[field2]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[field3]@[field2]>[peer_tag_param]
Call-ID: [call_id]
CSeq: [cseq] ACK
User-Agent: A2PC 10.3.1234
Content-Length: 0
]]>
</send>
<pause milliseconds="5000"/>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="/home/user1/TrialAndError/voicemail/dtmf_2833_1.pcap"/>
</action>
</nop>
<pause milliseconds="1500"/>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="/home/user1/TrialAndError/voicemail/dtmf_2833_1.pcap"/>
</action>
</nop>
<pause milliseconds="1500"/>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="/home/user1/TrialAndError/voicemail/dtmf_2833_1.pcap"/>
</action>
</nop>
<pause milliseconds="1500"/>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="/home/user1/TrialAndError/voicemail/dtmf_2833_1.pcap"/>
</action>
</nop>
<pause milliseconds="1500"/>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="/home/user1/TrialAndError/voicemail/dtmf_2833_1.pcap"/>
</action>
</nop>
<pause milliseconds="1500"/>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="/home/user1/TrialAndError/voicemail/dtmf_2833_pound.pcap"/>
</action>
</nop>
<pause milliseconds="9000"/>
<!-- This delay can be customized by the -d command-line option -->
<!-- or by adding a 'milliseconds = "value"' option here. -->
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="150">
<![CDATA[
BYE sip:[field3]@[field0]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "[field1]" <sip:[field1]@[field2]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[field3]@[field2]>[peer_tag_param]
Contact: sip:[field1]@[local_ip]:[local_port]
Call-ID: [call_id]
Max-Forwards: 70
CSeq: [cseq] BYE
User-Agent: A2PC 10.3.1234
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
</scenario>
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