It seems BSC is sending payload type GSM to LCR, but LCR send payload type PCMA on the sip channel.<div><br></div><div><div>07.03.12 22:45:03.907 CH(69): New call ref LCR<->BSC callref new=0x80000029</div><div>07.03.12 22:45:03.907 CH(69): Codec negotiation LCR<->BSC bearer capa='given by MS' speech version='AMR given' ignored='Not suitable for LCR' version='5 given' ignored='Not supported by LCR' version='EFR given' ignored='Not suitable for LCR' version='Full Rate given' version='Half Rate given' ignored='Not suitable for LCR'</div>
<div>07.03.12 22:45:03.908 CH(69): MNCC_SETUP_IND LCR<->BSC calling number=07777201 imsi=413011492012312 dialing number=4290080001</div><div>07.03.12 22:45:03.908 CH(69): MNCC_LCHAN_MODIFY LCR<->BSC speech version='Full Rate given' mode 0x01</div>
<div>07.03.12 22:45:03.908 CH(69): MNCC_CALL_PROC_REQ LCR<->BSC progress coding=3 location=1 descr=8</div><div>07.03.12 22:45:03.908 CH(69): unknown LCR<->BSC</div><div>07.03.12 22:45:03.908 CH(70): NEW handle handle new=0x8d65cc0</div>
<div>07.03.12 22:45:03.908 CH(70): INVITE from uri=<a href="mailto:sip%3A07777201@192.168.1.30">sip:07777201@192.168.1.30</a> to uri=<a href="http://sip:4290080001@192.168.1.25:4757">sip:4290080001@192.168.1.25:4757</a> rtp ip=103.10.172.30 port=30026,30027 payload=PCMA:8</div>
<div>07.03.12 22:45:03.930 CH(70): RESPOND respond value=183</div><div>07.03.12 22:45:03.930 CH(70): Payload received rtp payload=PCMA:8 payload=telephone-event:101</div><div>07.03.12 22:45:13.117 CH(69): MNCC_DISC_IND LCR<->BSC cause coding=3 location=0 value=16</div>
<div>07.03.12 22:45:13.148 CH(69): MNCC_REL_REQ LCR<->BSC</div><div>07.03.12 22:45:13.148 CH(70): CANCEL cause value=16</div><div>07.03.12 22:45:13.169 CH(70): RESPOND respond value=487</div></div><div><br></div><div>
<br><br><div class="gmail_quote">On Wed, Mar 7, 2012 at 3:43 PM, Nik Pakar <span dir="ltr"><<a href="mailto:nikpakar@gmail.com">nikpakar@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi Andreas, is the LCR actually transcoding gsm-fr to alaw towards sip side ?<div><br></div><div>Rgds</div><div><span class="HOEnZb"><font color="#888888">Nik</font></span><div><div class="h5"><br><br><div class="gmail_quote">
On Wed, Mar 7, 2012 at 3:17 PM, Nik Pakar <span dir="ltr"><<a href="mailto:nikpakar@gmail.com" target="_blank">nikpakar@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi Andreas,<div><br></div><div>Call signalling all works fine right through out from NITB to LCR and out on SIP. how ever im getting some strange media behaviour.</div>
<div><br></div><div>My setup is,</div><div><br></div><div>
[MS]---[nano.BTS]---[NITB/LCR]----[SIP Softswitch]</div><div><br></div><div>LCR is setup to bridge two interfaces, so what ever comes from gsm goes to sip and what ever comes from sip goes to gsm.</div><div><br></div><div>
Now a test call from a mobile to mobile, should go all the way to the softswitch and come back.</div><div><br></div><div>All works fine in terms of signalling.</div><div><br></div><div>But in media, LCR seems sending initial SDP to the softswitch as PCMA:8 not gsm FR.</div>
<div><br></div><div>So softswitch expect the media as PCMA and not transcoding.</div><div><br></div><div>Same if the call goes out from softswitch, still no medial as it think incoming media from LCR is on PCMA.</div><div>
<br></div><div>Any idea about this ?</div><div><br></div><div>This is the LCR trace -
<a href="http://pastebin.com/5PNKYc5m" target="_blank">http://pastebin.com/5PNKYc5m</a> </div><div>This is the sip trace from softswitch -
<a href="http://pastebin.com/cVtx1mFB" target="_blank">http://pastebin.com/cVtx1mFB</a> </div><div><br></div><div>Rgds</div><div><span><font color="#888888">Nik</font></span><div><div><br><br><div class="gmail_quote">
On Tue, Mar 6, 2012 at 3:38 PM, Nik Pakar <span dir="ltr"><<a href="mailto:nikpakar@gmail.com" target="_blank">nikpakar@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi Andreas, got it working both ways now. Many thanks for nice work.<div><br></div><div>I will now test it further with transcoding.</div>
<div><br>And start on the documentation.</div><div><br></div><div>Rgds</div><span><font color="#888888"><div>Nik</div></font></span><div><div>
<div><br><div class="gmail_quote">On Tue, Mar 6, 2012 at 12:51 PM, Nik Pakar <span dir="ltr"><<a href="mailto:nikpakar@gmail.com" target="_blank">nikpakar@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi Andreas, i now have calls coming from external sip->LCR->gsm<div><br></div><div>But still cant figure out the dial plan to send a call out on sip, gsm->LCR->sip</div><div><br></div><div>I tried,</div><div>
<br>
</div><div>dialing=072333444<span style="white-space:pre-wrap"> </span>: extern interfaces=sip prefix=072333444</div><div><br></div><div>But on LCR trace it shows as below. I think im missing some thing small here. Appreciate if you can give a little hint.</div>
<div><br></div><div>Thanks<br>nik</div><div><br></div><div><div>06.03.12 12:40:41.286 EP(1): ACTION (match) action goto line 11</div><div>06.03.12 12:40:41.286 EP(1): ACTION goto/menu (change to) ruleset extern dialing 072333444</div>
<div>06.03.12 12:40:41.286 EP(1): ACTION (match) action extern line 28</div><div>06.03.12 12:40:41.286 EP(1): ACTION extern (calling) number 072333444 interfaces sip</div><div>06.03.12 12:40:41.287 EP(1): SETUP ACKNOWLEDGE to CH(1)</div>
<div>06.03.12 12:40:41.287 EP(2): CHANNEL SELECTION (found given interface) interface sip</div><div>06.03.12 12:40:41.287 EP(2): INTERFACE (has no function) interface�@ </div><div>06.03.12 12:40:41.287 EP(2): INTERFACE (no free ports found)</div>
<div>06.03.12 12:40:41.287 EP(1): TONE to CH(1) directory default name cause_22</div><div>06.03.12 12:40:41.287 EP(1): DISCONNECT to CH(1) cause value=34 location=1-Local-PBX</div><div>06.03.12 12:40:41.287 CH(1): MNCC_DISC_REQ LCR<->BSC progress coding=3 location=1 descr=8 cause coding=3 location=1 value=34</div>
<div>06.03.12 12:40:56.246 CH(1): MNCC_REL_IND LCR<->BSC cause coding=3 location=0 value=16</div><div>06.03.12 12:40:56.247 EP(1): RELEASE from CH(1) cause value=16 location=0-User</div><div>06.03.12 12:40:56.247 EP(1): ACTION hangup</div>
<div><div>
<div><br></div><br><div class="gmail_quote">On Tue, Mar 6, 2012 at 7:44 AM, Nik Pakar <span dir="ltr"><<a href="mailto:nikpakar@gmail.com" target="_blank">nikpakar@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi Andreas,<div><br></div><div>It was apperently compiled on my debian while i had misdn libs isntalled. Now im trying on a fresh debian from the same set of sources which i got it working and without misdn, it fails to compile gsm.</div>
<div><br></div><div>Attached is my config output and compile break.</div><div><br></div><div><a href="http://pastebin.com/W6UHn4Lc" target="_blank">http://pastebin.com/W6UHn4Lc</a></div><div><br></div><div>So should i still install misdn even though its not used.</div>
<div><br></div><div>Rgds</div><div><span><font color="#888888">Nik</font></span><div><div><br><br><div class="gmail_quote">On Mon, Mar 5, 2012 at 10:05 AM, Andreas Eversberg <span dir="ltr"><<a href="mailto:andreas@eversberg.eu" target="_blank">andreas@eversberg.eu</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div>Alexander Chemeris wrote:<br>
> Does it mean that that now we can use LCR with other SIP<br>
> softswitches/PBX'es, like Freeswitch? I do not follow LCR development<br>
> closely, but that would be a very interesting development.<br>
><br>
</div>yes, this was my intention. gsm and sip interface of lcr will not rely<br>
on misdn anymore. currently i don't support audio transfer via chan_lcr,<br>
so chan_lcr will only work with isdn interfaces. the sip interface<br>
implementation has not much options, so it can only do sipmple<br>
point-to-point sip connections to a gateway or endpoint.<br>
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