# Asterisk -\> Softswitch -\> PSTN no ringing **URL:** \ **Category:** Asterisk Support\ **Created:** [November 8, 2013, 3:24am UTC](https://community.asterisk.org/t/asterisk-softswitch-pstn-no-ringing/44391 "2013-11-08T03:24:55Z")\ **Posts on this page:** 6\ **Page:** 1
**Author:** ![xblurone](https://avatars.discourse-cdn.com/v4/letter/x/97f17d/32.png) [@xblurone](https://community.asterisk.org/u/xblurone)\ **Post date:** [November 8, 2013, 3:24am UTC](https://community.asterisk.org/t/asterisk-softswitch-pstn-no-ringing/44391/1 "2013-11-08T03:24:55Z")
Hi, I have upgraded my asterisk to the latest 1.8.24 but now on outgoing calls I get no more ringing. My test: Grandstream GXP1405 -sip\> Asterisk -sip\> Voipswitch -sip\> Cisco AS5300 -\> ISDN = no ringing Grandstream GXP1405 -sip\> Voipswitch -sip\> Cisco AS5300 -\> ISDN = ringing (also announcements work) Note that when the destination answers the phone, voice works fine - so it’s not a NAT issue. I have tried everything with progressinband and premature media setting, to no avail. Where do I need to look ? My sip.conf at the moment looks as follows: progressinband=never prematuremedia=no transport=udp,tcp I am dialling out from an agi script with only the g parameter… $AGI-\>exec(‘Dial’,$channel); where $channel is something like “SIP/blahblah,70,g” as I want the script to continue after hangup. Any insight where I may have gone wrong is greatly appreciated. (incidentally reverting back to the old asterisk 1.8.15.0 where this worked, doesn’t work anymore, so I must have changed something else that I can’t remember) ---
**Author:** ![xblurone](https://avatars.discourse-cdn.com/v4/letter/x/97f17d/32.png) [@xblurone](https://community.asterisk.org/u/xblurone)\ **Post date:** [November 8, 2013, 1:38pm UTC](https://community.asterisk.org/t/asterisk-softswitch-pstn-no-ringing/44391/2 "2013-11-08T13:38:48Z")
Seems like I have figured it out… Asterisk now needs a Progress() call before the Dial command… After that the progress messages get passed through as well as early media. Is this normal ? I have also noticed that if one does Progress() more than once in the same call before the Dial command at some point the system will just deadlock - can still give commands, check peer status etc, but it won’t shutdown or accept any more registrations, or actually do anything… Have to kill -9 the asterisk process when that happens (1.8.24.0) ---
**Author:** ![xblurone](https://avatars.discourse-cdn.com/v4/letter/x/97f17d/32.png) [@xblurone](https://community.asterisk.org/u/xblurone)\ **Post date:** [November 8, 2013, 2:40pm UTC](https://community.asterisk.org/t/asterisk-softswitch-pstn-no-ringing/44391/3 "2013-11-08T14:40:23Z")
One more observation - I had to remove the Progress() command altogether on outgoing SIP-\>PSTN calls because the server would deadlock (with no idea how or where - probably in the sip channel) after processing about 100 calls or so… So, I’m back to square one… ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [November 8, 2013, 8:48pm UTC](https://community.asterisk.org/t/asterisk-softswitch-pstn-no-ringing/44391/4 "2013-11-08T20:48:18Z")
Please google “asterisk wiki backtrace” and follow the procedures for deadlocks. Assuming you are using a current version, it looks like you have a repeatable bug for which you need to submit a bug report. ---
**Author:** ![xblurone](https://avatars.discourse-cdn.com/v4/letter/x/97f17d/32.png) [@xblurone](https://community.asterisk.org/u/xblurone)\ **Post date:** [November 9, 2013, 9:55am UTC](https://community.asterisk.org/t/asterisk-softswitch-pstn-no-ringing/44391/5 "2013-11-09T09:55:22Z")
Hi, since this is a production server and I can’t seem to pin it down to a specific event (server can run for 6 hours and then “hang”, or 10 minutes and then “hang”) I have chosen to upgrade to 11.6.0 instead and see how that goes. First thing I noticed is that the CPU load is about 1/10th of what it was before… (about 2000 registered users) Let’s keep my fingers crossed. Should the same thing happen I know what to do and report it to the community. ---
**Author:** ![xblurone](https://avatars.discourse-cdn.com/v4/letter/x/97f17d/32.png) [@xblurone](https://community.asterisk.org/u/xblurone)\ **Post date:** [November 9, 2013, 1:32pm UTC](https://community.asterisk.org/t/asterisk-softswitch-pstn-no-ringing/44391/6 "2013-11-09T13:32:40Z")
Bad news - it still crashes - but I see a pattern - seems to be a DTMF issue ? These are the last messages logged in full.log in asterisk-11.6.0 before it stopped logging anything… [Nov 9 20:46:46] VERBOSE[5103][C-0000043c] app\_dial.c: – SIP/vps2-alaw-000007cd answered SIP/31611826-000007cc [Nov 9 20:46:46] VERBOSE[5103][C-0000043c] rtp\_engine.c: – Locally bridging SIP/31611826-000007cc and SIP/vps2-alaw-000007cd [Nov 9 20:46:47] VERBOSE[1165][C-00000389] netsock2.c: == Using UDPTL TOS bits 184 [Nov 9 20:46:47] VERBOSE[1165][C-00000389] netsock2.c: == Using UDPTL CoS mark 5 [Nov 9 20:47:59] DTMF[5103][C-0000043c] channel.c: DTMF end ‘8’ received on SIP/31611826-000007cc, duration 0 ms [Nov 9 20:47:59] DTMF[5103][C-0000043c] channel.c: DTMF begin emulation of ‘8’ with duration 100 queued on SIP/31611826-000007cc [Nov 9 20:47:59] DTMF[5103][C-0000043c] channel.c: DTMF end emulation of ‘8’ queued on SIP/31611826-000007cc [Nov 9 20:48:25] DTMF[5095][C-0000043a] channel.c: DTMF end ‘D’ received on SIP/31612298-000007c8, duration 0 ms [Nov 9 20:48:25] DTMF[5095][C-0000043a] channel.c: DTMF begin emulation of ‘D’ with duration 100 queued on SIP/31612298-000007c8 [Nov 9 20:48:25] DTMF[5095][C-0000043a] channel.c: DTMF end emulation of ‘D’ queued on SIP/31612298-000007c8 [Nov 9 20:50:46] DTMF[5103][C-0000043c] channel.c: DTMF end ‘4’ received on SIP/31611826-000007cc, duration 0 ms [Nov 9 20:50:46] DTMF[5103][C-0000043c] channel.c: DTMF begin emulation of ‘4’ with duration 100 queued on SIP/31611826-000007cc [Nov 9 20:50:47] DTMF[5103][C-0000043c] channel.c: DTMF end emulation of ‘4’ queued on SIP/31611826-000007cc [Nov 9 20:51:15] DTMF[5095][C-0000043a] channel.c: DTMF end ‘5’ received on SIP/31612298-000007c8, duration 0 ms [Nov 9 20:51:15] DTMF[5095][C-0000043a] channel.c: DTMF begin emulation of ‘5’ with duration 100 queued on SIP/31612298-000007c8 [Nov 9 20:51:16] DTMF[5095][C-0000043a] channel.c: DTMF end emulation of ‘5’ queued on SIP/31612298-000007c8 [Nov 9 20:51:16] DTMF[5095][C-0000043a] channel.c: DTMF end ‘C’ received on SIP/31612298-000007c8, duration 0 ms [Nov 9 20:51:16] DTMF[5095][C-0000043a] channel.c: DTMF begin emulation of ‘C’ with duration 100 queued on SIP/31612298-000007c8 [Nov 9 20:51:16] DTMF[5095][C-0000043a] channel.c: DTMF end emulation of ‘C’ queued on SIP/31612298-000007c8 [Nov 9 20:54:39] DTMF[5103][C-0000043c] channel.c: DTMF end ‘B’ received on SIP/vps2-alaw-000007cd, duration 0 ms [Nov 9 20:54:39] DTMF[5103][C-0000043c] channel.c: DTMF begin emulation of ‘B’ with duration 100 queued on SIP/vps2-alaw-000007cd [Nov 9 20:54:39] DTMF[5103][C-0000043c] channel.c: DTMF end emulation of ‘B’ queued on SIP/vps2-alaw-000007cd [Nov 9 20:58:55] DTMF[5103][C-0000043c] channel.c: DTMF end ‘3’ received on SIP/vps2-alaw-000007cd, duration 0 ms [Nov 9 20:58:55] DTMF[5103][C-0000043c] channel.c: DTMF begin emulation of ‘3’ with duration 100 queued on SIP/vps2-alaw-000007cd [Nov 9 20:58:55] DTMF[5103][C-0000043c] channel.c: DTMF end emulation of ‘3’ queued on SIP/vps2-alaw-000007cd [Nov 9 20:59:21] DTMF[5103][C-0000043c] channel.c: DTMF end ‘1’ received on SIP/vps2-alaw-000007cd, duration 0 ms [Nov 9 20:59:21] DTMF[5103][C-0000043c] channel.c: DTMF begin emulation of ‘1’ with duration 100 queued on SIP/vps2-alaw-000007cd [Nov 9 20:59:21] DTMF[5103][C-0000043c] channel.c: DTMF end emulation of ‘1’ queued on SIP/vps2-alaw-000007cd [Nov 9 20:59:51] DTMF[5103][C-0000043c] channel.c: DTMF end ‘6’ received on SIP/31611826-000007cc, duration 0 ms [Nov 9 20:59:51] DTMF[5103][C-0000043c] channel.c: DTMF begin emulation of ‘6’ with duration 100 queued on SIP/31611826-000007cc [Nov 9 20:59:51] DTMF[5103][C-0000043c] channel.c: DTMF end emulation of ‘6’ queued on SIP/31611826-000007cc [Nov 9 21:00:25] DTMF[4567][C-00000388] channel.c: DTMF end ‘6’ received on SIP/vps2-alaw-00000690, duration 0 ms [Nov 9 21:00:25] DTMF[4567][C-00000388] channel.c: DTMF begin emulation of ‘6’ with duration 100 queued on SIP/vps2-alaw-00000690 [Nov 9 21:00:25] DTMF[4567][C-00000388] channel.c: DTMF end emulation of ‘6’ queued on SIP/vps2-alaw-00000690 [Nov 9 21:01:06] DTMF[5103][C-0000043c] channel.c: DTMF end ‘1’ received on SIP/vps2-alaw-000007cd, duration 0 ms [Nov 9 21:01:06] DTMF[5103][C-0000043c] channel.c: DTMF begin emulation of ‘1’ with duration 100 queued on SIP/vps2-alaw-000007cd [Nov 9 21:01:06] DTMF[5103][C-0000043c] channel.c: DTMF end emulation of ‘1’ queued on SIP/vps2-alaw-000007cd [Nov 9 21:01:13] DTMF[5103][C-0000043c] channel.c: DTMF end ‘4’ received on SIP/vps2-alaw-000007cd, duration 0 ms [Nov 9 21:01:13] DTMF[5103][C-0000043c] channel.c: DTMF begin emulation of ‘4’ with duration 100 queued on SIP/vps2-alaw-000007cd [Nov 9 21:01:14] DTMF[5103][C-0000043c] channel.c: DTMF end emulation of ‘4’ queued on SIP/vps2-alaw-000007cd [Nov 9 21:02:56] DTMF[5103][C-0000043c] channel.c: DTMF end ‘5’ received on SIP/31611826-000007cc, duration 0 ms [Nov 9 21:02:56] DTMF[5103][C-0000043c] channel.c: DTMF begin emulation of ‘5’ with duration 100 queued on SIP/31611826-000007cc [Nov 9 21:02:56] DTMF[5103][C-0000043c] channel.c: DTMF end emulation of ‘5’ queued on SIP/31611826-000007cc