search for: sig_analog

...I-Express) chan_dahdi.conf contains: usecallerid = yes cidsignalling=dtmf cidstart=polarity_in Signalling is fxsks. Log (calling number is 9811066XXX): [Sep 14 08:21:11] DEBUG[9337]: chan_dahdi.c:11895 do_monitor: Monitor doohicky got event Ring Begin on channel 1 [Sep 14 08:21:11] DEBUG[9337]: sig_analog.c:3621 analog_handle_init_event: channel (1) - signaling (5) - event (ANALOG_EVENT_RINGBEGIN) [Sep 14 08:21:11] DEBUG[9337]: chan_dahdi.c:11895 do_monitor: Monitor doohicky got event Ring/Answered on channel 1 [Sep 14 08:21:11] DEBUG[9337]: sig_analog.c:3621 analog_handle_init_event: channel (1)...

...e one span over one night which was working fine on the old box. Once plugged in there were no alarms, Sangoma wanpipemon utility showed "connected". I tried calling in on a DID number, and in the 'full' log, with debug and verbose set to 100: [Dec 5 00:51:37] VERBOSE[5283] sig_analog.c: -- Starting simple switch on 'DAHDI/9-1' [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=1.17E+04, Et=1.45E+06, s/n= 0.01 [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=2.76E+03, Et=1.10E+06, s/n= 0.00 [Dec 5 00:51:38] DEBUG[5283] dsp.c: tone 1100, Ew=2.06E+04,...

...ot;no call pickup possible" message as soon as I press the '8': [Feb 12 18:17:39] -- Starting simple switch on 'DAHDI/2-1' [Feb 12 18:17:39] DEBUG[2339]: devicestate.c:442 devstate_event: device 'DAHDI/2' state '2' [Feb 12 18:17:42] DEBUG[5898][C-0000000e]: sig_analog.c:1600 analog_handle_dtmf: Begin DTMF digit: 0x2A '*' on DAHDI/2-1 [Feb 12 18:17:42] DEBUG[5898][C-0000000e]: chan_dahdi.c:2145 my_handle_dtmf: Begin DTMF digit: 0x2A '*' on DAHDI/2-1 [Feb 12 18:17:42] DEBUG[5898][C-0000000e]: sig_analog.c:1600 analog_handle_dtmf: End DTMF digit: 0x...

...ix for loop declarations - alembic: Fix quoting of the 100rel column - pbx.c: Fix gcc 12 compiler warning. - app_audiosocket: Fixed timeout with -1 to avoid busy loop. - download_externals: Fix a few version related issues - main/refer.c: Fix double free in refer_data_destructor + potential leak - sig_analog: Add Called Subscriber Held capability. - app_macro: Fix locking around datastore access - Revert "app_stack: Print proper exit location for PBXless channels." - .github: Use generic releaser - install_prereq: Fix dependency install on aarch64. - res_pjsip.c: Set contact_user on incoming...

...ix for loop declarations - alembic: Fix quoting of the 100rel column - pbx.c: Fix gcc 12 compiler warning. - app_audiosocket: Fixed timeout with -1 to avoid busy loop. - download_externals: Fix a few version related issues - main/refer.c: Fix double free in refer_data_destructor + potential leak - sig_analog: Add Called Subscriber Held capability. - app_macro: Fix locking around datastore access - Revert "app_stack: Print proper exit location for PBXless channels." - .github: Use generic releaser - install_prereq: Fix dependency install on aarch64. - res_pjsip.c: Set contact_user on incoming...

...ix for loop declarations - alembic: Fix quoting of the 100rel column - pbx.c: Fix gcc 12 compiler warning. - app_audiosocket: Fixed timeout with -1 to avoid busy loop. - download_externals: Fix a few version related issues - main/refer.c: Fix double free in refer_data_destructor + potential leak - sig_analog: Add Called Subscriber Held capability. - app_macro: Fix locking around datastore access - Revert "app_stack: Print proper exit location for PBXless channels." - .github: Use generic releaser - install_prereq: Fix dependency install on aarch64. - res_pjsip.c: Set contact_user on incoming...

...ix for loop declarations - alembic: Fix quoting of the 100rel column - pbx.c: Fix gcc 12 compiler warning. - app_audiosocket: Fixed timeout with -1 to avoid busy loop. - download_externals: Fix a few version related issues - main/refer.c: Fix double free in refer_data_destructor + potential leak - sig_analog: Add Called Subscriber Held capability. - app_macro: Fix locking around datastore access - Revert "app_stack: Print proper exit location for PBXless channels." - .github: Use generic releaser - install_prereq: Fix dependency install on aarch64. - res_pjsip.c: Set contact_user on incoming...

...0 card, and set that channel to have the context "doorphone". The problem is that the extension is never executed. With debugging on, I see: Nov 23 16:37:23] DEBUG[20077]: chan_dahdi.c:11896 do_monitor: Monitor doohicky got event Ring Begin on channel 12 [Nov 23 16:37:23] DEBUG[20077]: sig_analog.c:3621 analog_handle_init_event: channel (12) - signaling (5) - event (ANALOG_EVENT_RINGBEGIN) [Nov 23 16:37:24] DEBUG[20077]: chan_dahdi.c:11896 do_monitor: Monitor doohicky got event Ring/Answered on channel 12 [Nov 23 16:37:24] DEBUG[20077]: sig_analog.c:3621 analog_handle_init_event: channel (1...

Hi everyone! I have a Digium Card TDM410 But, it appear for me this massege chan_dahdi.c:8061 handle_alarms: Detected alarm on channel 3: Red Alarm But my line is ok! But sometimes it back sig_analog.c:3807 analog_handle_init_event: Alarm cleared on channel 2 But it again back to red alarm. What can be happen? My lines is all ok! But when I put on Digium Card TDM410 is very inconsistent Thanks

...llowme: fix issue with enable_callee_prompt=no (#88) - sounds: Update download URL to use HTTPS. - configure: Makefile downloader enable follow redirects. - res_musiconhold: Add option to loop last file. - chan_dahdi: Fix Caller ID presentation for FXO ports. - AMI: Add CoreShowChannelMap action. - sig_analog: Add fuller Caller ID support. - res_stasis.c: Add new type 'sdp_label' for bridge creation. - app_queue: Preserve reason for realtime queues - .github: Fix issues with cherry-pick-reminder - indications: logging changes - .github Ignore error when adding reviewrs to PR - .github: Update fi...

...llowme: fix issue with enable_callee_prompt=no (#88) - sounds: Update download URL to use HTTPS. - configure: Makefile downloader enable follow redirects. - res_musiconhold: Add option to loop last file. - chan_dahdi: Fix Caller ID presentation for FXO ports. - AMI: Add CoreShowChannelMap action. - sig_analog: Add fuller Caller ID support. - res_stasis.c: Add new type 'sdp_label' for bridge creation. - app_queue: Preserve reason for realtime queues - .github: Fix issues with cherry-pick-reminder - indications: logging changes - .github Ignore error when adding reviewrs to PR - .github: Update fi...

...llowme: fix issue with enable_callee_prompt=no (#88) - sounds: Update download URL to use HTTPS. - configure: Makefile downloader enable follow redirects. - res_musiconhold: Add option to loop last file. - chan_dahdi: Fix Caller ID presentation for FXO ports. - AMI: Add CoreShowChannelMap action. - sig_analog: Add fuller Caller ID support. - res_stasis.c: Add new type 'sdp_label' for bridge creation. - app_queue: Preserve reason for realtime queues - .github: Fix issues with cherry-pick-reminder - indications: logging changes - .github Ignore error when adding reviewrs to PR - .github: Update fi...

...llowme: fix issue with enable_callee_prompt=no (#88) - sounds: Update download URL to use HTTPS. - configure: Makefile downloader enable follow redirects. - res_musiconhold: Add option to loop last file. - chan_dahdi: Fix Caller ID presentation for FXO ports. - AMI: Add CoreShowChannelMap action. - sig_analog: Add fuller Caller ID support. - res_stasis.c: Add new type 'sdp_label' for bridge creation. - app_queue: Preserve reason for realtime queues - .github: Fix issues with cherry-pick-reminder - indications: logging changes - .github Ignore error when adding reviewrs to PR - .github: Update fi...

...t; Hi everyone! >> >> I have a Digium Card TDM410 >> >> But, it appear for me this massege >> >> chan_dahdi.c:8061 handle_alarms: Detected alarm on channel 3: Red Alarm >> >> But my line is ok! >> >> But sometimes it back >> >> sig_analog.c:3807 analog_handle_init_event: Alarm cleared on channel 2 >> >> But it again back to red alarm. >> >> What can be happen? >> >> My lines is all ok! But when I put on Digium Card TDM410 is very >> inconsistent >> >> Thanks >> >> --...

...re's a debug from a 911 call. [Jan 31 11:29:53] DEBUG[9876][C-00000005]: pbx.c:4890 pbx_extension_helper: Launching 'Dial' -- Executing [s at InFromSIP:1] Dial("SIP/SIP-00000005", "DAHDI/1/2177772001") in new stack [Jan 31 11:29:53] DEBUG[9876][C-00000005]: sig_analog.c:820 analog_available: analog_available 1 [Jan 31 11:29:53] DEBUG[9876][C-00000005]: sig_analog.c:845 analog_available: Channel 1 off hook, can't use [Jan 31 11:29:53] WARNING[9876][C-00000005]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 17 - User...

...ix for loop declarations - alembic: Fix quoting of the 100rel column - pbx.c: Fix gcc 12 compiler warning. - app_audiosocket: Fixed timeout with -1 to avoid busy loop. - download_externals: Fix a few version related issues - main/refer.c: Fix double free in refer_data_destructor + potential leak - sig_analog: Add Called Subscriber Held capability. - Revert "app_stack: Print proper exit location for PBXless channels." - install_prereq: Fix dependency install on aarch64. - res_pjsip.c: Set contact_user on incoming call local Contact header - extconfig: Allow explicit DB result set ordering to b...

...ix for loop declarations - alembic: Fix quoting of the 100rel column - pbx.c: Fix gcc 12 compiler warning. - app_audiosocket: Fixed timeout with -1 to avoid busy loop. - download_externals: Fix a few version related issues - main/refer.c: Fix double free in refer_data_destructor + potential leak - sig_analog: Add Called Subscriber Held capability. - Revert "app_stack: Print proper exit location for PBXless channels." - install_prereq: Fix dependency install on aarch64. - res_pjsip.c: Set contact_user on incoming call local Contact header - extconfig: Allow explicit DB result set ordering to b...

Good morning, I have a digium wctdm24xxp in my asterisk box, i am not able to see the callerid when the call is incoming from the fxo line, i live in Brazil, how can i change the signaling from fsk to dtmf? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL:

...every few days, my POTS line gets stuck. The other ports are working for intercom, but attempts to place outbound calls through the POTS line, whether from a SIP phone or one of the DAHDI extensions, get a fast busy, and this appears in the Asterisk log: [Apr 5 21:29:35] WARNING[8191][C-0000003e] sig_analog.c: Ring/Off-hook in strange state 6 on channel 4 I have to restart asterisk and/or dahdi to get it working again. Can someone tell me what this message means and suggest how I might prevent this from happening? Is my hardware flaky, or is there something I need to update in my configuration? Tha...

...he transferee channel when it is transferred via a native transfer mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by Mark Michelson (license 5049) * Restore call progress code for analog ports. Extracting sig_analog from chan_dahdi lost call progress detection functionality. Fix analog ports from considering a call answered immediately after dialing has completed if the callprogress option is enabled. (closes issue ASTERISK-18841) Reported by: Richard Miller Patched by Richard Miller * Fix regre...