Re: [asterisk-ss7] No audio using libss7 over E1 (SOLVED)
On Fri, Aug 3, 2012 at 1:06 AM, Krzysztof Drewicz krzysztofdrew...@gmail.com wrote: 2012/8/2 Marcelo Pacheco marc...@m2j.com.br No, THATS A BAD SS7 EXPERIENCE. You must learn this BEFORE trying to mess around with ISUP. And the very 1st posts suggested - to pull cable out. But, nowadays, you have guys playing with ss7 network that could not even make proper E1 loop-back plug... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-ss7 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-ss7 -- IT DESK www.essencekey.com i...@essencekey.com Contact: +91-1164722856 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-ss7 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-ss7
Re: [asterisk-ss7] No audio using libss7 over E1 (SOLVED)
FINALLY after many hours and much hair pulling, emailing and assistance from this asterisk-ss7 list we've finally got it working!! It turns out that the E1 links were swapped around on the provider's side which means that CICs were swapped around.. A BIG thank you to everyone that responded. Thank you for your assistance and guidance. I've definitely had a good SS7 learning experience! :) Get important Linux and industry-related news at: facebook.com/dcdata http://facebook.com/dcdata Kind regards, David Wilson CNS,CLS, LINUX+, CLA, DCTS, LPIC3 *LinuxTech CC t/a DcData* CK number: 2001/058368/23 *Website:* http://www.dcdata.co.za *Support:* +27(0)860-1-LINUX *Mobile:* +27(0)824147413 *Tel:* +27(0)333446100 *Fax:* +27(0)866878971 On 08/02/2012 02:07 PM, David Wilson wrote: OK. An inbound call was placed to our system which landed on CIC1. / //Unhandled optional parameter 0x8 'Optional forward call indicator' [0x0 ] Unhandled optional parameter 0x31 'Propagation Delay Counter' [0x0 0x5a ] Unhandled optional parameter 0x3a 'Unknown' [0x44 0x2 0x68 0x0 0x0 0x0 ] Unhandled optional parameter 0x3f 'Location Number' [0x84 0x93 0x62 0x78 0x6 0x0 0x0 0x0 ] Unhandled optional parameter 0x39 'Parameter Compatibility Information' [0x31 0xd0 0x3a 0xd0 0x3f 0xc0 ] -- Accepting call to '01900' on CIC 1 -- Executing [01900@from-ss7:1] Answer(DAHDI/1-1, ) in new stack -- Executing [01900@from-ss7:2] Monitor(DAHDI/1-1, wav,1343909018.3,m) in new stack -- Executing [01900@from-ss7:3] Playback(DAHDI/1-1, /var/lib/asterisk/sounds/en/followme/pls-hold-while-try) in new stack -- DAHDI/1-1 Playing '/var/lib/asterisk/sounds/en/followme/pls-hold-while-try.gsm' (language 'en') -- Executing [01900@from-ss7:4] MusicOnHold(DAHDI/1-1, default,60) in new stack -- Started music on hold, class 'default', on DAHDI/1-1 -- Stopped music on hold on DAHDI/1-1/ I ran dahdi_monitor 1 -vv and could see my music on hold traffic being transmitted as under (TX). I've tried downgrading asterisk to asterisk-1.6.0 as suggested by some posts but I still get the same result. The calling in person does not hear my music on hold. Get important Linux and industry-related news at: facebook.com/dcdata http://facebook.com/dcdata Kind regards, David Wilson CNS,CLS, LINUX+, CLA, DCTS, LPIC3 *LinuxTech CC t/a DcData* CK number: 2001/058368/23 *Website:* http://www.dcdata.co.za *Support:* +27(0)860-1-LINUX *Mobile:* +27(0)824147413 *Tel:* +27(0)333446100 *Fax:* +27(0)866878971 On 08/02/2012 01:34 PM, David Wilson wrote: Thank you for your reply Kalolyan. I've made the adjustments that you've suggested and will retest shortly. A quick check though with regards to dahdi_monitor syntax. May I use dahdi_monitor as follows to show the CIC? /dahdi_monitor 1 -v/ For monitoring channel 1. Get important Linux and industry-related news at: facebook.com/dcdata http://facebook.com/dcdata Kind regards, David Wilson CNS,CLS, LINUX+, CLA, DCTS, LPIC3 *LinuxTech CC t/a DcData* CK number: 2001/058368/23 *Website:* http://www.dcdata.co.za *Support:* +27(0)860-1-LINUX *Mobile:* +27(0)824147413 *Tel:* +27(0)333446100 *Fax:* +27(0)866878971 On 08/02/2012 01:02 PM, Kaloyan Kovachev wrote: Hi, On Thu, 02 Aug 2012 12:19:06 +0200, David Wilsond...@dcdata.co.za wrote: */etc/asterisk/chan_dahdi.conf*: [trunkgroups] [channels] usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A102 port 1 [slot:4 bus:12 span:1]wanpipe1 switchtype=euroisdn you should not define switchtype with ss7 - it is only for PRI signaling context=from-ss7 group=0 echocancel=no signaling=ss7 linkset=1 ss7type=itu ss7_called_nai=dynamic ss7_calling_nai=dynamic ss7_internationalprefix=00 ss7_nationalprefix=0 ss7_subscriberprefix= ss7_unknownprefix= ;ss7_explicitacm=yes pointcode=1379 adjpointcode=1288 defaultdpc=1288 networkindicator=national ;First E1 sigchan=16 cicbeginswith=1 channel=1-15 cicbeginswith=17 channel=17-31 ;Second E1 sigchan=47 cicbeginswith=33 channel=32-46 cicbeginswith=49 channel=48-62 please define your sigchans last. you may use dahdi_monitor to check on which CIC you are receiving the audio from the other side and if it is not the one on which you are sending it - you can be sure about the misalignment. You may comment out the second E1 and check with the telco if they get the signaling on their first link or (as you can't swap the cables) just swap them in your config and see if the audio is working: ;First E1 cicbeginswith=1 channel=32-46 cicbeginswith=17 channel=48-62 ;Second E1 cicbeginswith=33 channel=1-15 cicbeginswith=49 channel=17-31 sigchan=16 sigchan=47 */etc/asterisk/extensions.conf*
Re: [asterisk-ss7] No audio using libss7 over E1 (SOLVED)
No, THATS A BAD SS7 EXPERIENCE. You must learn this BEFORE trying to mess around with ISUP. ISUP is serious stuff, do it right. If you haven't spent at least 2 full days studying ISUP and SS7, than you're not even close to ready to mess around with ISUP in production. On 08/02/12 14:38, David Wilson wrote: FINALLY after many hours and much hair pulling, emailing and assistance from this asterisk-ss7 list we've finally got it working!! It turns out that the E1 links were swapped around on the provider's side which means that CICs were swapped around.. A BIG thank you to everyone that responded. Thank you for your assistance and guidance. I've definitely had a good SS7 learning experience! :) Get important Linux and industry-related news at: facebook.com/dcdata http://facebook.com/dcdata Kind regards, David Wilson CNS,CLS, LINUX+, CLA, DCTS, LPIC3 *LinuxTech CC t/a DcData* CK number: 2001/058368/23 *Website:* http://www.dcdata.co.za *Support:* +27(0)860-1-LINUX *Mobile:* +27(0)824147413 *Tel:* +27(0)333446100 *Fax:* +27(0)866878971 On 08/02/2012 02:07 PM, David Wilson wrote: OK. An inbound call was placed to our system which landed on CIC1. / //Unhandled optional parameter 0x8 'Optional forward call indicator' [0x0 ] Unhandled optional parameter 0x31 'Propagation Delay Counter' [0x0 0x5a ] Unhandled optional parameter 0x3a 'Unknown' [0x44 0x2 0x68 0x0 0x0 0x0 ] Unhandled optional parameter 0x3f 'Location Number' [0x84 0x93 0x62 0x78 0x6 0x0 0x0 0x0 ] Unhandled optional parameter 0x39 'Parameter Compatibility Information' [0x31 0xd0 0x3a 0xd0 0x3f 0xc0 ] -- Accepting call to '01900' on CIC 1 -- Executing [01900@from-ss7:1] Answer(DAHDI/1-1, ) in new stack -- Executing [01900@from-ss7:2] Monitor(DAHDI/1-1, wav,1343909018.3,m) in new stack -- Executing [01900@from-ss7:3] Playback(DAHDI/1-1, /var/lib/asterisk/sounds/en/followme/pls-hold-while-try) in new stack -- DAHDI/1-1 Playing '/var/lib/asterisk/sounds/en/followme/pls-hold-while-try.gsm' (language 'en') -- Executing [01900@from-ss7:4] MusicOnHold(DAHDI/1-1, default,60) in new stack -- Started music on hold, class 'default', on DAHDI/1-1 -- Stopped music on hold on DAHDI/1-1/ I ran dahdi_monitor 1 -vv and could see my music on hold traffic being transmitted as under (TX). I've tried downgrading asterisk to asterisk-1.6.0 as suggested by some posts but I still get the same result. The calling in person does not hear my music on hold. Get important Linux and industry-related news at: facebook.com/dcdata http://facebook.com/dcdata Kind regards, David Wilson CNS,CLS, LINUX+, CLA, DCTS, LPIC3 *LinuxTech CC t/a DcData* CK number: 2001/058368/23 *Website:* http://www.dcdata.co.za *Support:* +27(0)860-1-LINUX *Mobile:* +27(0)824147413 *Tel:* +27(0)333446100 *Fax:* +27(0)866878971 On 08/02/2012 01:34 PM, David Wilson wrote: Thank you for your reply Kalolyan. I've made the adjustments that you've suggested and will retest shortly. A quick check though with regards to dahdi_monitor syntax. May I use dahdi_monitor as follows to show the CIC? /dahdi_monitor 1 -v/ For monitoring channel 1. Get important Linux and industry-related news at: facebook.com/dcdata http://facebook.com/dcdata Kind regards, David Wilson CNS,CLS, LINUX+, CLA, DCTS, LPIC3 *LinuxTech CC t/a DcData* CK number: 2001/058368/23 *Website:* http://www.dcdata.co.za *Support:* +27(0)860-1-LINUX *Mobile:* +27(0)824147413 *Tel:* +27(0)333446100 *Fax:* +27(0)866878971 On 08/02/2012 01:02 PM, Kaloyan Kovachev wrote: Hi, On Thu, 02 Aug 2012 12:19:06 +0200, David Wilsond...@dcdata.co.za wrote: */etc/asterisk/chan_dahdi.conf*: [trunkgroups] [channels] usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A102 port 1 [slot:4 bus:12 span:1] wanpipe1 switchtype=euroisdn you should not define switchtype with ss7 - it is only for PRI signaling context=from-ss7 group=0 echocancel=no signaling=ss7 linkset=1 ss7type=itu ss7_called_nai=dynamic ss7_calling_nai=dynamic ss7_internationalprefix=00 ss7_nationalprefix=0 ss7_subscriberprefix= ss7_unknownprefix= ;ss7_explicitacm=yes pointcode=1379 adjpointcode=1288 defaultdpc=1288 networkindicator=national ;First E1 sigchan=16 cicbeginswith=1 channel=1-15 cicbeginswith=17 channel=17-31 ;Second E1 sigchan=47 cicbeginswith=33 channel=32-46 cicbeginswith=49 channel=48-62 please define your sigchans last. you may use dahdi_monitor to check on which CIC you are receiving the audio from the other side and if it is not the one on which you are sending it - you can be sure about the misalignment. You may comment out the second E1 and check with the telco if they get the signaling on their
Re: [asterisk-ss7] No audio using libss7 over E1 (SOLVED)
2012/8/2 Marcelo Pacheco marc...@m2j.com.br No, THATS A BAD SS7 EXPERIENCE. You must learn this BEFORE trying to mess around with ISUP. And the very 1st posts suggested - to pull cable out. But, nowadays, you have guys playing with ss7 network that could not even make proper E1 loop-back plug... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-ss7 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-ss7
