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Alexander Chemeris alexander.chemeris at gmail.comNik, You could check whether this is PCMA or GSM-FR with RTP stream bitrate. Capture the stream with Wireshark and look at RTP payload size or use a voice call information dialog (Telephony->VoIP Calls). PCMA bitrate is 64kbit (often 160bytes per 20ms), while bitrate GSM-FR is 13.2kbit (usually 33bytes per 20ms). On Wed, Mar 7, 2012 at 21:36, Nik Pakar <nikpakar at gmail.com> wrote: > It seems BSC is sending payload type GSM to LCR, but LCR send payload type > PCMA on the sip channel. > > 07.03.12 22:45:03.907 CH(69): New call ref LCR<->BSC callref new=0x80000029 > 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' > 07.03.12 22:45:03.908 CH(69): MNCC_SETUP_IND LCR<->BSC calling > number=07777201 imsi=413011492012312 dialing number=4290080001 > 07.03.12 22:45:03.908 CH(69): MNCC_LCHAN_MODIFY LCR<->BSC speech > version='Full Rate given' mode 0x01 > 07.03.12 22:45:03.908 CH(69): MNCC_CALL_PROC_REQ LCR<->BSC progress > coding=3 location=1 descr=8 > 07.03.12 22:45:03.908 CH(69): unknown LCR<->BSC > 07.03.12 22:45:03.908 CH(70): NEW handle handle new=0x8d65cc0 > 07.03.12 22:45:03.908 CH(70): INVITE from uri=sip:07777201 at 192.168.1.30 to > uri=sip:4290080001 at 192.168.1.25:4757 rtp ip=103.10.172.30 port=30026,30027 > payload=PCMA:8 > 07.03.12 22:45:03.930 CH(70): RESPOND respond value=183 > 07.03.12 22:45:03.930 CH(70): Payload received rtp payload=PCMA:8 > payload=telephone-event:101 > 07.03.12 22:45:13.117 CH(69): MNCC_DISC_IND LCR<->BSC cause coding=3 > location=0 value=16 > 07.03.12 22:45:13.148 CH(69): MNCC_REL_REQ LCR<->BSC > 07.03.12 22:45:13.148 CH(70): CANCEL cause value=16 > 07.03.12 22:45:13.169 CH(70): RESPOND respond value=487 > > > > On Wed, Mar 7, 2012 at 3:43 PM, Nik Pakar <nikpakar at gmail.com> wrote: >> >> Hi Andreas, is the LCR actually transcoding gsm-fr to alaw towards sip >> side ? >> >> Rgds >> Nik >> >> >> On Wed, Mar 7, 2012 at 3:17 PM, Nik Pakar <nikpakar at gmail.com> wrote: >>> >>> Hi Andreas, >>> >>> Call signalling all works fine right through out from NITB to LCR and out >>> on SIP. how ever im getting some strange media behaviour. >>> >>> My setup is, >>> >>> [MS]---[nano.BTS]---[NITB/LCR]----[SIP Softswitch] >>> >>> 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. >>> >>> Now a test call from a mobile to mobile, should go all the way to the >>> softswitch and come back. >>> >>> All works fine in terms of signalling. >>> >>> But in media, LCR seems sending initial SDP to the softswitch as PCMA:8 >>> not gsm FR. >>> >>> So softswitch expect the media as PCMA and not transcoding. >>> >>> Same if the call goes out from softswitch, still no medial as it think >>> incoming media from LCR is on PCMA. >>> >>> Any idea about this ? >>> >>> This is the LCR trace - http://pastebin.com/5PNKYc5m >>> This is the sip trace from softswitch - http://pastebin.com/cVtx1mFB >>> >>> Rgds >>> Nik >>> >>> >>> On Tue, Mar 6, 2012 at 3:38 PM, Nik Pakar <nikpakar at gmail.com> wrote: >>>> >>>> Hi Andreas, got it working both ways now. Many thanks for nice work. >>>> >>>> I will now test it further with transcoding. >>>> >>>> And start on the documentation. >>>> >>>> Rgds >>>> Nik >>>> >>>> On Tue, Mar 6, 2012 at 12:51 PM, Nik Pakar <nikpakar at gmail.com> wrote: >>>>> >>>>> Hi Andreas, i now have calls coming from external sip->LCR->gsm >>>>> >>>>> But still cant figure out the dial plan to send a call out on sip, >>>>> gsm->LCR->sip >>>>> >>>>> I tried, >>>>> >>>>> dialing=072333444 : extern interfaces=sip prefix=072333444 >>>>> >>>>> But on LCR trace it shows as below. I think im missing some thing small >>>>> here. Appreciate if you can give a little hint. >>>>> >>>>> Thanks >>>>> nik >>>>> >>>>> 06.03.12 12:40:41.286 EP(1): ACTION (match) action goto line 11 >>>>> 06.03.12 12:40:41.286 EP(1): ACTION goto/menu (change to) ruleset >>>>> extern dialing 072333444 >>>>> 06.03.12 12:40:41.286 EP(1): ACTION (match) action extern line 28 >>>>> 06.03.12 12:40:41.286 EP(1): ACTION extern (calling) number 072333444 >>>>> interfaces sip >>>>> 06.03.12 12:40:41.287 EP(1): SETUP ACKNOWLEDGE to CH(1) >>>>> 06.03.12 12:40:41.287 EP(2): CHANNEL SELECTION (found given interface) >>>>> interface sip >>>>> 06.03.12 12:40:41.287 EP(2): INTERFACE (has no function) interface�@ >>>>> 06.03.12 12:40:41.287 EP(2): INTERFACE (no free ports found) >>>>> 06.03.12 12:40:41.287 EP(1): TONE to CH(1) directory default name >>>>> cause_22 >>>>> 06.03.12 12:40:41.287 EP(1): DISCONNECT to CH(1) cause value=34 >>>>> location=1-Local-PBX >>>>> 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 >>>>> 06.03.12 12:40:56.246 CH(1): MNCC_REL_IND LCR<->BSC cause coding=3 >>>>> location=0 value=16 >>>>> 06.03.12 12:40:56.247 EP(1): RELEASE from CH(1) cause value=16 >>>>> location=0-User >>>>> 06.03.12 12:40:56.247 EP(1): ACTION hangup >>>>> >>>>> >>>>> On Tue, Mar 6, 2012 at 7:44 AM, Nik Pakar <nikpakar at gmail.com> wrote: >>>>>> >>>>>> Hi Andreas, >>>>>> >>>>>> 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. >>>>>> >>>>>> Attached is my config output and compile break. >>>>>> >>>>>> http://pastebin.com/W6UHn4Lc >>>>>> >>>>>> So should i still install misdn even though its not used. >>>>>> >>>>>> Rgds >>>>>> Nik >>>>>> >>>>>> >>>>>> On Mon, Mar 5, 2012 at 10:05 AM, Andreas Eversberg >>>>>> <andreas at eversberg.eu> wrote: >>>>>>> >>>>>>> Alexander Chemeris wrote: >>>>>>> > Does it mean that that now we can use LCR with other SIP >>>>>>> > softswitches/PBX'es, like Freeswitch? I do not follow LCR >>>>>>> > development >>>>>>> > closely, but that would be a very interesting development. >>>>>>> > >>>>>>> yes, this was my intention. gsm and sip interface of lcr will not >>>>>>> rely >>>>>>> on misdn anymore. currently i don't support audio transfer via >>>>>>> chan_lcr, >>>>>>> so chan_lcr will only work with isdn interfaces. the sip interface >>>>>>> implementation has not much options, so it can only do sipmple >>>>>>> point-to-point sip connections to a gateway or endpoint. >>>>>> >>>>>> >>>>> >>>> >>> >> > -- Regards, Alexander Chemeris.