HELP - timeout to GSM not working


#1

I am trying to setup a “follow-me” extension using a combination of SIP phones and GSM. This extension will be used as a hotline for customer support. If the SIP phones are not answered, then the next SIP extension is tried, after all the in-house extensions are exhausted, then Asterisk should begin to ring customer support engineers’ mobile phones.

The config below is a test setup:

; Test Follow-me
exten => 312,1,Dial(SIP/312,15,r)
exten => 312,2,Dial(SIP/451,15,r)
exten => 312,3,Dial(zap/g1/01721043273,20,r)
exten => 312,4,Dial(zap/g1/01723456891,20,r)
exten => 312,5,VoiceMail,u212
exten => 312,101,VoiceMail,b212

Everything works as expected, except when the SIP phones go unanswered, and the GSM phone begins to ring. From here the timeout option just does not work. The first GSM number will just ring and ring until the mobile users’ voice mail picks up.

If anyone has any experience getting this to perform the way it should I would appreciate any help.

Regards,
Joe


#2

I am going through the same thing too. I configured as above to ring my remote phone if no response but it instead hangs up right after it says “please wait to connect your call”. I have the incoming context of my extensions.conf set as:

[from-pstn]
exten=>s,1,Ringing
exten=>s,2,Answer
exten=>s,3,Wait,2
exten=>s,4,Background(vm-enter-num-to-call)
exten=>s,5,Wait,2
;exten=>1,1,Dial(SIP/gw1|30) ;Pickupgroup 1
;exten=>1,2,Voicemail,ugw1
exten=>1,1,Dial(SIP/200&SIP/201&SIP/202&SIP/203&sSIP/204/SIP/205&SIP/206&SIP/207&SIP/208/SIP/209&SIP/210|30)
;All Home Tech Solutions extensions
;exten=>1,2,playback(pls-wait-connect-call)
;exten=>1,3,Setvar(NewCaller=${CALLERIDNUM})
;exten=>1,4,SetCIDNum(0${CALLERIDNUM})
;exten=>1,5,dial(${TRUNK}c/mycellnumberhere,20,r)
;exten=>1,6,SetCIDNum(${NewCaller})
;exten=>1,7,voicemail2(u200@default)
;exten=>1,101,voicemail2(b200@default)
;exten=>1,102,hangup

exten=>1,2,playback(pls-wait-connect-call)
exten=>1,3,dial(SIP/sip.broadvoice.com/mycellnumberhere,30)
exten=>1,4,voicemail2(u200@default)
exten=>1,5,hangup

…and the debug displays as follows:

May 29 14:21:09 DEBUG[2591] chan_sip.c: Setting NAT on RTP to 524288
May 29 14:21:09 DEBUG[2591] chan_sip.c: Checking SIP call limits for device myBVnumberhere
May 29 14:21:09 DEBUG[2591] chan_sip.c: build_route: Contact hop:
May 29 14:21:09 VERBOSE[32270] logger.c: – Executing Goto(“SIP/myBVnumberhere-7a2e”, “from-pstn|s|1”) in new stack
May 29 14:21:09 VERBOSE[32270] logger.c: – Goto (from-pstn,s,1)
May 29 14:21:09 VERBOSE[32270] logger.c: – Executing Ringing(“SIP/myBVnumberhere-7a2e”, “”) in new stack
May 29 14:21:09 VERBOSE[32270] logger.c: – Executing Answer(“SIP/myBVnumberhere-7a2e”, “”) in new stack
May 29 14:21:09 VERBOSE[32270] logger.c: – Executing Wait(“SIP/myBVnumberhere-7a2e”, “2”) in new stack
May 29 14:21:09 DEBUG[2591] chan_sip.c: Stopping retransmission on ‘1ca0090-ca@147.135.12.128’ of Response 1: Match Found
May 29 14:21:11 VERBOSE[32270] logger.c: – Executing BackGround(“SIP/myBVnumberhere-7a2e”, “vm-enter-num-to-call”) in new stack
May 29 14:21:11 DEBUG[32270] channel.c: Scheduling timer at 160 sample intervals
May 29 14:21:11 VERBOSE[32270] logger.c: – Playing ‘vm-enter-num-to-call’ (language ‘en’)
May 29 14:21:13 DEBUG[32270] chan_sip.c: * Detected inband DTMF '1’
May 29 14:21:13 DEBUG[32270] channel.c: Scheduling timer at 0 sample intervals
May 29 14:21:13 DEBUG[32270] pbx.c: Oooh, got something to jump out with (‘1’)!
May 29 14:21:13 VERBOSE[32270] logger.c: == CDR updated on SIP/myBVnumberhere-7a2e
May 29 14:21:13 VERBOSE[32270] logger.c: – Executing Dial(“SIP/myBVnumberhere-7a2e”, “SIP/200&SIP/201&SIP/202&SIP/203&SIP/204/SIP/205&SIP/206&SIP/207&SIP/208/SIP/209&SIP/210|30”) in new stack
May 29 14:21:13 DEBUG[32270] chan_sip.c: Setting NAT on RTP to 0
May 29 14:21:13 DEBUG[32270] chan_sip.c: Outgoing Call for 200
May 29 14:21:13 VERBOSE[32270] logger.c: – Called 200
May 29 14:21:13 NOTICE[32270] app_dial.c: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
May 29 14:21:13 DEBUG[32270] chan_sip.c: Setting NAT on RTP to 0
May 29 14:21:13 DEBUG[32270] chan_sip.c: Outgoing Call for 202
May 29 14:21:13 VERBOSE[32270] logger.c: – Called 202
May 29 14:21:13 NOTICE[32270] app_dial.c: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
May 29 14:21:13 NOTICE[32270] app_dial.c: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
May 29 14:21:13 NOTICE[32270] app_dial.c: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
May 29 14:21:13 NOTICE[32270] app_dial.c: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
May 29 14:21:13 NOTICE[32270] app_dial.c: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
May 29 14:21:13 NOTICE[32270] app_dial.c: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
May 29 14:21:13 DEBUG[2591] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on ‘055ffa544b498a8459180a4350762d00@192.168.1.12’ Request 102: Found
May 29 14:21:13 DEBUG[2591] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on ‘75eb0cc60b86800776ed0e1d6d7bee23@192.168.1.12’ Request 102: Found
May 29 14:21:13 DEBUG[2591] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on ‘055ffa544b498a8459180a4350762d00@192.168.1.12’ Request 102: Found
May 29 14:21:13 VERBOSE[32270] logger.c: – SIP/200-cdff is ringing
May 29 14:21:13 DEBUG[32270] channel.c: Driver for channel ‘SIP/myBVnumberhere-7a2e’ does not support indication 3, emulating it
May 29 14:21:13 DEBUG[32270] channel.c: Scheduling timer at 160 sample intervals
May 29 14:21:13 DEBUG[32270] channel.c: Generator got voice, switching to phase locked mode
May 29 14:21:13 DEBUG[32270] channel.c: Scheduling timer at 0 sample intervals
May 29 14:21:13 DEBUG[2591] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on ‘75eb0cc60b86800776ed0e1d6d7bee23@192.168.1.12’ Request 102: Found
May 29 14:21:13 VERBOSE[32270] logger.c: – SIP/202-befb is ringing
May 29 14:21:18 DEBUG[2591] chan_sip.c: Scheduled a registration timeout for sip.broadvoice.com id #73370
May 29 14:21:18 DEBUG[2591] chan_sip.c: Stopping retransmission on ‘1f04e4b93cef3aff386b37b457012684@127.0.0.1’ of Request 104: Match Found
May 29 14:21:18 DEBUG[2591] chan_sip.c: Registration successful
May 29 14:21:18 DEBUG[2591] chan_sip.c: Cancelling timeout 73370
May 29 14:21:26 DEBUG[2591] chan_sip.c: Auto destroying call '1f04e4b93cef3aff386b37b457012684@127.0.0.1’
May 29 14:21:26 DEBUG[2591] chan_sip.c: Stopping retransmission on ‘3b537f911afd9cb3388e46b531c88b4e@192.168.1.12’ of Request 102: Match Found
May 29 14:21:29 DEBUG[2591] chan_sip.c: Stopping retransmission on ‘6b1af5151da667cc1d12265642ac358b@192.168.1.12’ of Request 102: Match Found
May 29 14:21:34 DEBUG[2591] chan_sip.c: Auto destroying call 'F4B137E5D2D143C8989287F367455311@192.168.1.12’
May 29 14:21:42 DEBUG[2591] chan_sip.c: Scheduled a registration timeout for sip.broadvoice.com id #73382
May 29 14:21:42 DEBUG[2591] chan_sip.c: Stopping retransmission on ‘1f04e4b93cef3aff386b37b457012684@127.0.0.1’ of Request 105: Match Found
May 29 14:21:42 DEBUG[2591] chan_sip.c: Registration successful
May 29 14:21:42 DEBUG[2591] chan_sip.c: Cancelling timeout 73382
May 29 14:21:44 VERBOSE[32270] logger.c: – Nobody picked up in 30000 ms
May 29 14:21:44 DEBUG[32270] channel.c: Scheduling timer at 0 sample intervals
May 29 14:21:44 DEBUG[32270] chan_sip.c: update_call_counter(202) - decrement call limit counter
May 29 14:21:44 DEBUG[32270] chan_sip.c: Acked pending invite 102
May 29 14:21:44 DEBUG[32270] chan_sip.c: Stopping retransmission on ‘75eb0cc60b86800776ed0e1d6d7bee23@192.168.1.12’ of Request 102: Match Found
May 29 14:21:44 DEBUG[32270] chan_sip.c: update_call_counter(200) - decrement call limit counter
May 29 14:21:44 DEBUG[32270] chan_sip.c: Acked pending invite 102
May 29 14:21:44 DEBUG[32270] chan_sip.c: Stopping retransmission on ‘055ffa544b498a8459180a4350762d00@192.168.1.12’ of Request 102: Match Found
May 29 14:21:44 DEBUG[32270] app_dial.c: Exiting with DIALSTATUS=NOANSWER.
May 29 14:21:44 VERBOSE[32270] logger.c: – Executing Playback(“SIP/myBVnumberhere-7a2e”, “pls-wait-connect-call”) in new stack
May 29 14:21:44 DEBUG[32270] channel.c: Scheduling timer at 160 sample intervals
May 29 14:21:44 VERBOSE[32270] logger.c: – Playing ‘pls-wait-connect-call’ (language ‘en’)
May 29 14:21:44 DEBUG[2591] chan_sip.c: Stopping retransmission on ‘055ffa544b498a8459180a4350762d00@192.168.1.12’ of Request 102: Match Not Found
May 29 14:21:44 DEBUG[2591] chan_sip.c: Stopping retransmission on ‘055ffa544b498a8459180a4350762d00@192.168.1.12’ of Request 102: Match Found
May 29 14:21:44 DEBUG[2591] chan_sip.c: Stopping retransmission on ‘75eb0cc60b86800776ed0e1d6d7bee23@192.168.1.12’ of Request 102: Match Not Found
May 29 14:21:44 DEBUG[2591] chan_sip.c: Stopping retransmission on ‘75eb0cc60b86800776ed0e1d6d7bee23@192.168.1.12’ of Request 102: Match Found
May 29 14:21:46 DEBUG[32270] channel.c: Scheduling timer at 0 sample intervals
May 29 14:21:46 DEBUG[32270] channel.c: Scheduling timer at 0 sample intervals
May 29 14:21:46 VERBOSE[32270] logger.c: – Executing Dial(“SIP/myBVnumberhere-7a2e”, “SIP/sip.broadvoice.com/remotenumberhere|30”) in new stack
May 29 14:21:46 DEBUG[32270] chan_sip.c: Outgoing Call for remotenumberhere
May 29 14:21:46 VERBOSE[32270] logger.c: – Called sip.broadvoice.com/remotenumberhere
May 29 14:21:46 DEBUG[2591] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on ‘3aa0086f53f1d125299500d504982ad9@192.168.1.12’ Request 102: Found
May 29 14:21:46 DEBUG[2591] chan_sip.c: Acked pending invite 102
May 29 14:21:46 DEBUG[2591] chan_sip.c: Stopping retransmission on ‘3aa0086f53f1d125299500d504982ad9@192.168.1.12’ of Request 102: Match Found
May 29 14:21:46 WARNING[2591] chan_sip.c: Forbidden - wrong password on authentication for INVITE to ‘“Washington DC” ;tag=as4cbe3e94’
May 29 14:21:46 VERBOSE[32270] logger.c: – SIP/sip.broadvoice.com-df61 is circuit-busy
May 29 14:21:46 DEBUG[32270] chan_sip.c: update_call_counter(remotenumberhere) - decrement call limit counter
May 29 14:21:46 VERBOSE[32270] logger.c: == Everyone is busy/congested at this time (1:0/1/0)
May 29 14:21:46 DEBUG[32270] app_dial.c: Exiting with DIALSTATUS=CONGESTION.
May 29 14:21:46 WARNING[32270] pbx.c: No application ‘voicemail2’ for extension (from-pstn, 1, 4)
May 29 14:21:46 VERBOSE[32270] logger.c: == Spawn extension (from-pstn, 1, 4) exited non-zero on ‘SIP/myBVnumberhere-7a2e’
May 29 14:21:46 DEBUG[32270] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
May 29 14:21:46 DEBUG[32270] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid) VALUES (‘2006-05-29 14:21:09’,’“Washington DC” ‘,‘cellnumberhere’,‘1’,‘from-pstn’, ‘SIP/myBVnumberhere-7a2e’,‘SIP/sip.broadvoice.com-df61’,‘Dial’,‘SIP/sip.broadvoice.com/remotenumberhere|30’,37,37,‘ANSWERED’,3,’’,‘1148926869.238’)
May 29 14:21:46 DEBUG[32270] chan_sip.c: update_call_counter(myBVnumberhere) - decrement call limit counter
May 29 14:21:46 DEBUG[2591] chan_sip.c: Stopping retransmission on ‘1ca0090-ca@147.135.12.128’ of Request 102: Match Found
May 29 14:21:50 DEBUG[2591] chan_sip.c: Auto destroying call '1f04e4b93cef3aff386b37b457012684@127.0.0.1’
May 29 14:22:06 DEBUG[2591] chan_sip.c: Scheduled a registration timeout for sip.broadvoice.com id #73392
May 29 14:22:06 DEBUG[2591] chan_sip.c: Stopping retransmission on ‘1f04e4b93cef3aff386b37b457012684@127.0.0.1’ of Request 106: Match Found
May 29 14:22:06 DEBUG[2591] chan_sip.c: Registration successful
May 29 14:22:06 DEBUG[2591] chan_sip.c: Cancelling timeout 73392
May 29 14:22:06 DEBUG[2591] chan_sip.c: Cancelling timeout 73392


#3

As I just posted above, I suspect either BV (broadvoice) is enforcing call limit on my account (has anyone else been able to make such multiple simultaneous calls) or there is some error somewhere that I cannot seem to pinpoint.


#4

In addition to my last posting, after analyzing the debug, I noticed the lines where a dial attempt to reach the remote number was initiated but uncompleted for some wierd reason that I am still yet to figure out:

May 29 14:21:46 VERBOSE[32270] logger.c: – Executing Dial(“SIP/myBVnumberhere-7a2e”, “SIP/sip.broadvoice.com/remotenumberhere|30”) in new stack
May 29 14:21:46 DEBUG[32270] chan_sip.c: Outgoing Call for remotenumberhere
May 29 14:21:46 VERBOSE[32270] logger.c: – Called sip.broadvoice.com/remotenumberhere —remote dial attempt initializing here
May 29 14:21:46 DEBUG[2591] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on ‘3aa0086f53f1d125299500d504982ad9@192.168.1.12’ Request 102: Found
May 29 14:21:46 DEBUG[2591] chan_sip.c: Acked pending invite 102
May 29 14:21:46 DEBUG[2591] chan_sip.c: Stopping retransmission on ‘3aa0086f53f1d125299500d504982ad9@192.168.1.12’ of Request 102: Match Found
May 29 14:21:46 WARNING[2591] chan_sip.c: Forbidden - wrong password on authentication for INVITE to '“Washington DC” ;tag=as4cbe3e94’
May 29 14:21:46 VERBOSE[32270] logger.c: – SIP/sip.broadvoice.com-df61 is circuit-busy
May 29 14:21:46 DEBUG[32270] chan_sip.c: update_call_counter(remotenumberhere) - decrement call limit counter
May 29 14:21:46 VERBOSE[32270] logger.c: == Everyone is busy/congested at this time (1:0/1/0)
May 29 14:21:46 DEBUG[32270] app_dial.c: Exiting with DIALSTATUS=CONGESTION.