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SIPp
late-offer uas->uac reinvite
Hello, i need to implement the following scenario:
UAT--->INVITE (sdp)--->SIPp
UAT<--- 100 <---SIPp
UAT<--- 183 <---SIPp
UAT<--- 200 OK <---SIPp
UAT----> ACK --->SIPp
UAT<--- INVITE (NO sdp) <---SIPp
UAT---> 100 --->SIPp
UAT---> 183 --->SIPp
UAT---> 200 OK (sdp) -->SIPp
UAT<--- ACK (sdp) <---SIPp
UAT---> BYE -->SIPp
UAT<--- ACK <--SIPp
And i can't figure this out, this is what i have so far, but the UAT on re-invite responds with "Not Here":
this is the scenario file:
<?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 'uas' scenario. -->
<!-- -->
<scenario name="Basic UAS responder">
<!-- 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 request="INVITE" crlf="true">
</recv>
<!-- The '[last_*]' keyword is replaced automatically by the -->
<!-- specified header if it was present in the last message received -->
<!-- (except if it was a retransmission). If the header was not -->
<!-- present or if no message has been received, the '[last_*]' -->
<!-- keyword is discarded, and all bytes until the end of the line -->
<!-- are also discarded. -->
<!-- -->
<!-- If the specified header was present several times in the -->
<!-- message, all occurences are concatenated (CRLF seperated) -->
<!-- to be used in place of the '[last_*]' keyword. -->
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
]]>
</send>
<recv request="ACK"
rtd="true"
crlf="true">
</recv>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="g711a.pcap"/>
</action>
</nop>
<!-- Pause 90 seconds, which is approximately the duration of the -->
<!-- PCAP file -->
<pause milliseconds="5000"/>
<send retrans="500">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
CSeq: 1 INVITE
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<!-- Pause 90 seconds, which is approximately the duration of the -->
<!-- PCAP file -->
<pause milliseconds="5000"/>
<recv request="BYE">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the BYE again. -->
<timewait milliseconds="4000"/>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
here's the log:
----------------------------------------------- 2022-09-14 14:48:57.611122
UDP message received [1475] bytes :
INVITE sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:172.18.2.142;lr>
Via: SIP/2.0/UDP 172.18.2.142:5060;branch=z9hG4bK000d.4d1ac5b26acb1c80bf5d4a503c05db40.0
Via: SIP/2.0/UDP 172.18.2.142:5088;received=172.18.2.142;rport=5088;branch=z9hG4bKBe691ejQvg1gN
Max-Forwards: 69
From: <sip:[email protected]>;tag=BDyrX323rct6c
To: <sip:[email protected]:5060>
Call-ID: 7085bce1-aece-123b-e7af-4cd98fcab23a
CSeq: 57026076 INVITE
Contact: <sip:[email protected]:5088>
User-Agent: FreeSWITCH-mod_sofia/1.10.6-release+git~20210624T121012Z~eb2252ba97~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 469
X-FS-Support: update_display,send_info
Remote-Party-ID: <sip:[email protected]>;party=calling;screen=yes;privacy=off
X-ID: hx-e6679729f5e05f3f9a908d9ac8664b65
v=0
o=FreeSWITCH 1663146553 1663146554 IN IP4 172.18.2.142
s=FreeSWITCH
c=IN IP4 172.18.2.142
t=0 0
m=audio 13184 RTP/AVP 9 8 0 3 101 13
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
m=video 14502 RTP/AVP 102
b=AS:1024
a=rtpmap:102 VP8/90000
a=rtcp-fb:102 ccm fir
a=rtcp-fb:102 ccm tmmbr
a=rtcp-fb:102 nack
a=rtcp-fb:102 nack pli
----------------------------------------------- 2022-09-14 14:48:57.611493
UDP message sent (456 bytes):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.18.2.142:5060;branch=z9hG4bK000d.4d1ac5b26acb1c80bf5d4a503c05db40.0, SIP/2.0/UDP 172.18.2.142:5088;received=172.18.2.142;rport=5088;branch=z9hG4bKBe691ejQvg1gN
From: <sip:[email protected]>;tag=BDyrX323rct6c
To: <sip:[email protected]:5060>;tag=28756SIPpTag011
Call-ID: 7085bce1-aece-123b-e7af-4cd98fcab23a
CSeq: 57026076 INVITE
Contact: <sip:172.18.2.142:5090;transport=UDP>
Content-Length: 0
----------------------------------------------- 2022-09-14 14:48:57.612667
UDP message sent (645 bytes):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.2.142:5060;branch=z9hG4bK000d.4d1ac5b26acb1c80bf5d4a503c05db40.0, SIP/2.0/UDP 172.18.2.142:5088;received=172.18.2.142;rport=5088;branch=z9hG4bKBe691ejQvg1gN
From: <sip:[email protected]>;tag=BDyrX323rct6c
To: <sip:[email protected]:5060>;tag=28756SIPpTag011
Call-ID: 7085bce1-aece-123b-e7af-4cd98fcab23a
CSeq: 57026076 INVITE
Contact: <sip:172.18.2.142:5090;transport=UDP>
Content-Type: application/sdp
Content-Length: 159
v=0
o=user1 53655765 2353687637 IN IP4 172.18.2.142
s=-
c=IN IP4 172.18.2.142
t=0 0
m=audio 6000 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
----------------------------------------------- 2022-09-14 14:48:57.615039
UDP message received [383] bytes :
ACK sip:172.18.2.142:5090;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.18.2.142:5088;rport;branch=z9hG4bKcQZ2392tSSQ3g
Max-Forwards: 70
From: <sip:[email protected]>;tag=BDyrX323rct6c
To: <sip:[email protected]:5060>;tag=28756SIPpTag011
Call-ID: 7085bce1-aece-123b-e7af-4cd98fcab23a
CSeq: 57026076 ACK
Contact: <sip:[email protected]:5088>
Content-Length: 0
----------------------------------------------- 2022-09-14 14:49:02.618281
UDP message sent (599 bytes):
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.2.142:5088;rport;branch=z9hG4bKcQZ2392tSSQ3g
From: <sip:[email protected]>;tag=BDyrX323rct6c
To: <sip:[email protected]:5060>;tag=28756SIPpTag011;tag=28756SIPpTag011
Call-ID: 7085bce1-aece-123b-e7af-4cd98fcab23a
CSeq: 1 INVITE
Contact: <sip:172.18.2.142:5090;transport=UDP>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: 135
v=0
o=user1 53655765 2353687637 IN IP4 172.18.2.142
s=-
c=IN IP4 172.18.2.142
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
----------------------------------------------- 2022-09-14 14:49:02.619757
UDP message received [421] bytes :
SIP/2.0 404 Not here
Via: SIP/2.0/UDP 172.18.2.142:5088;rport=5090;branch=z9hG4bKcQZ2392tSSQ3g;received=172.18.2.142
From: <sip:[email protected]>;tag=BDyrX323rct6c
To: <sip:[email protected]:5060>;tag=28756SIPpTag011;tag=28756SIPpTag011
Call-ID: 7085bce1-aece-123b-e7af-4cd98fcab23a
CSeq: 1 INVITE
X-ID: hx-e6679729f5e05f3f9a908d9ac8664b65
Server: kamailio (5.0.2 (x86_64/linux))
Content-Length: 0
-----------------------------------------------
Unexpected UDP message received:
SIP/2.0 404 Not here
Via: SIP/2.0/UDP 172.18.2.142:5088;rport=5090;branch=z9hG4bKcQZ2392tSSQ3g;received=172.18.2.142
From: <sip:[email protected]>;tag=BDyrX323rct6c
To: <sip:[email protected]:5060>;tag=28756SIPpTag011;tag=28756SIPpTag011
Call-ID: 7085bce1-aece-123b-e7af-4cd98fcab23a
CSeq: 1 INVITE
X-ID: hx-e6679729f5e05f3f9a908d9ac8664b65
Server: kamailio (5.0.2 (x86_64/linux))
Content-Length: 0
SIPp's command:
sipp -sf uas-invite-multiple-codecs.xml -p 5090 -t un -max_socket 1024 -trace_err -error_file error_file.log -trace_msg -message_file message_file.log -i 172.18.2.142 -s 361110000 172.18.2.142:5060
In your re-invite:
- you're using the same VIA tag as the ACK which is not correct
- You have a duplicated TO tag
Try to correct these 2 first.
@sissime thanks, you're right. still,
UDP message received [1474] bytes :
INVITE sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:172.18.2.142;lr>
Via: SIP/2.0/UDP 172.18.2.142:5060;branch=z9hG4bK439.f40b80e3d9c71fb247958473931c8d3f.0
Via: SIP/2.0/UDP 172.18.2.142:5088;received=172.18.2.142;rport=5088;branch=z9hG4bK9vvHQtB15UX1g
Max-Forwards: 69
From: <sip:[email protected]>;tag=v6ByUF1Z73m4H
To: <sip:[email protected]:5060>
Call-ID: cd64bcfb-af6e-123b-e7af-4cd98fcab23a
CSeq: 57060514 INVITE
Contact: <sip:[email protected]:5088>
User-Agent: FreeSWITCH-mod_sofia/1.10.6-release+git~20210624T121012Z~eb2252ba97~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 469
X-FS-Support: update_display,send_info
Remote-Party-ID: <sip:[email protected]>;party=calling;screen=yes;privacy=off
X-ID: hx-8bbcf9e1983eef2fd8e13048768a2cee
v=0
o=FreeSWITCH 1663213792 1663213793 IN IP4 172.18.2.142
s=FreeSWITCH
c=IN IP4 172.18.2.142
t=0 0
m=audio 14820 RTP/AVP 9 8 0 3 101 13
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
m=video 11058 RTP/AVP 102
b=AS:1024
a=rtpmap:102 VP8/90000
a=rtcp-fb:102 ccm fir
a=rtcp-fb:102 ccm tmmbr
a=rtcp-fb:102 nack
a=rtcp-fb:102 nack pli
----------------------------------------------- 2022-09-15 09:56:52.901304
UDP message sent (455 bytes):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.18.2.142:5060;branch=z9hG4bK439.f40b80e3d9c71fb247958473931c8d3f.0, SIP/2.0/UDP 172.18.2.142:5088;received=172.18.2.142;rport=5088;branch=z9hG4bK9vvHQtB15UX1g
From: <sip:[email protected]>;tag=v6ByUF1Z73m4H
To: <sip:[email protected]:5060>;tag=21120SIPpTag011
Call-ID: cd64bcfb-af6e-123b-e7af-4cd98fcab23a
CSeq: 57060514 INVITE
Contact: <sip:172.18.2.142:5090;transport=UDP>
Content-Length: 0
----------------------------------------------- 2022-09-15 09:56:52.902483
UDP message sent (644 bytes):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.2.142:5060;branch=z9hG4bK439.f40b80e3d9c71fb247958473931c8d3f.0, SIP/2.0/UDP 172.18.2.142:5088;received=172.18.2.142;rport=5088;branch=z9hG4bK9vvHQtB15UX1g
From: <sip:[email protected]>;tag=v6ByUF1Z73m4H
To: <sip:[email protected]:5060>;tag=21120SIPpTag011
Call-ID: cd64bcfb-af6e-123b-e7af-4cd98fcab23a
CSeq: 57060514 INVITE
Contact: <sip:172.18.2.142:5090;transport=UDP>
Content-Type: application/sdp
Content-Length: 159
v=0
o=user1 53655765 2353687637 IN IP4 172.18.2.142
s=-
c=IN IP4 172.18.2.142
t=0 0
m=audio 6000 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
----------------------------------------------- 2022-09-15 09:56:52.903845
UDP message received [383] bytes :
ACK sip:172.18.2.142:5090;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.18.2.142:5088;rport;branch=z9hG4bKa6NaSNv424Kmc
Max-Forwards: 70
From: <sip:[email protected]>;tag=v6ByUF1Z73m4H
To: <sip:[email protected]:5060>;tag=21120SIPpTag011
Call-ID: cd64bcfb-af6e-123b-e7af-4cd98fcab23a
CSeq: 57060514 ACK
Contact: <sip:[email protected]:5088>
Content-Length: 0
----------------------------------------------- 2022-09-15 09:56:57.909673
UDP message sent (570 bytes):
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.2.142:5090;branch=z9hG4bK-21120-1-6
From: <sip:[email protected]>;tag=v6ByUF1Z73m4H
To: <sip:[email protected]:5060>;tag=21120SIPpTag011
Call-ID: cd64bcfb-af6e-123b-e7af-4cd98fcab23a
CSeq: 1 INVITE
Contact: <sip:172.18.2.142:5090;transport=UDP>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: 135
v=0
o=user1 53655765 2353687637 IN IP4 172.18.2.142
s=-
c=IN IP4 172.18.2.142
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
----------------------------------------------- 2022-09-15 09:56:57.911022
UDP message received [365] bytes :
SIP/2.0 404 Not here
Via: SIP/2.0/UDP 172.18.2.142:5090;branch=z9hG4bK-21120-1-6
From: <sip:[email protected]>;tag=v6ByUF1Z73m4H
To: <sip:[email protected]:5060>;tag=21120SIPpTag011
Call-ID: cd64bcfb-af6e-123b-e7af-4cd98fcab23a
CSeq: 1 INVITE
X-ID: hx-8bbcf9e1983eef2fd8e13048768a2cee
Server: kamailio (5.0.2 (x86_64/linux))
Content-Length: 0
-----------------------------------------------
Unexpected UDP message received:
SIP/2.0 404 Not here
Via: SIP/2.0/UDP 172.18.2.142:5090;branch=z9hG4bK-21120-1-6
From: <sip:[email protected]>;tag=v6ByUF1Z73m4H
To: <sip:[email protected]:5060>;tag=21120SIPpTag011
Call-ID: cd64bcfb-af6e-123b-e7af-4cd98fcab23a
CSeq: 1 INVITE
X-ID: hx-8bbcf9e1983eef2fd8e13048768a2cee
Server: kamailio (5.0.2 (x86_64/linux))
Content-Length: 0
my scenario:
<?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 'uas' scenario. -->
<!-- -->
<scenario name="Basic UAS responder">
<!-- 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 request="INVITE" crlf="true">
</recv>
<!-- The '[last_*]' keyword is replaced automatically by the -->
<!-- specified header if it was present in the last message received -->
<!-- (except if it was a retransmission). If the header was not -->
<!-- present or if no message has been received, the '[last_*]' -->
<!-- keyword is discarded, and all bytes until the end of the line -->
<!-- are also discarded. -->
<!-- -->
<!-- If the specified header was present several times in the -->
<!-- message, all occurences are concatenated (CRLF seperated) -->
<!-- to be used in place of the '[last_*]' keyword. -->
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
]]>
</send>
<recv request="ACK"
rtd="true"
crlf="true">
</recv>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="g711a.pcap"/>
</action>
</nop>
<!-- Pause 90 seconds, which is approximately the duration of the -->
<!-- PCAP file -->
<pause milliseconds="5000"/>
<send retrans="500">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
[last_From:]
[last_To:]
[last_Call-ID:]
CSeq: 1 INVITE
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<!-- Pause 90 seconds, which is approximately the duration of the -->
<!-- PCAP file -->
<!-- <pause milliseconds="5000"/> -->
<recv request="BYE">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the BYE again. -->
<timewait milliseconds="4000"/>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
I really appreciate your help!
I think it's not working because i'm not sending the branch in the reINVITE, but i'm trying to retrieve it with a regexp and i can't figure it out:
<ereg regexp="^Via.*branch=(.*)" search_in="hdr" header="Via:" check_it="true" assign_to="2" />
<log message="Branch=<[$2]>"/>
Again thanks!