No outbound "all-circuits-busy-now&pls-try-call-later"

Distro Discussion & Help
1

Hi,

I have a problem with my outgoing calls, “outbound” only gets all-circuits-busy-now&pls-try-call-later message, incoming calls work with no problem even the IVR is working very nice, I have tried all different way’s of dial rules and just can’t get it to work.

My version of FreePBX 2.10.0.8
My version of Asterisk (Ver. 1.8.10.0)

I know I have a lot of errors but I don’t know how to fix them, I’m using audio codec G729a but it gives me frame.c: Cannot allow unknown format ‘g729a’, if I remove the “a” then I have no audio on my phones, not even the voice mail, but put it back and audio is back. ?

any help will be appreciated. Thank you…

[2012-06-21 00:14:53] VERBOSE[28312] pbx.c: -- Executing [[email protected]:3] NoOp("SIP/102-0000004f", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to other trunks") in new stack
[2012-06-21 00:14:53] VERBOSE[28312] pbx.c: -- Executing [[email protected]:4] Set("SIP/102-0000004f", "CALLERID(number)=102") in new stack
[2012-06-21 00:14:53] VERBOSE[28312] pbx.c: -- Executing [[email protected]:6] Macro("SIP/102-0000004f", "outisbusy,") in new stack
[2012-06-21 00:14:53] VERBOSE[28312] pbx.c: -- Executing [[email protected]:1] Progress("SIP/102-0000004f", "") in new stack
[2012-06-21 00:14:53] VERBOSE[28312] pbx.c: -- Executing [[email protected]:2] GotoIf("SIP/102-0000004f", "0?emergency,1") in new stack
[2012-06-21 00:14:53] VERBOSE[28312] pbx.c: -- Executing [[email protected]:3] GotoIf("SIP/102-0000004f", "0?intracompany,1") in new stack
[2012-06-21 00:14:53] VERBOSE[28312] pbx.c: -- Executing [[email protected]:4] Playback("SIP/102-0000004f", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
[2012-06-21 00:14:53] VERBOSE[28312] file.c: -- <SIP/102-0000004f> Playing 'all-circuits-busy-now.gsm' (language 'en')
[2012-06-21 00:14:55] VERBOSE[28312] file.c: -- <SIP/102-0000004f> Playing 'pls-try-call-later.gsm' (language 'en')
[2012-06-21 00:14:57] VERBOSE[28312] pbx.c: -- Executing [[email protected]:5] Congestion("SIP/102-0000004f", "20") in new stack
[2012-06-21 00:14:57] WARNING[28312] channel.c: Prodding channel 'SIP/102-0000004f' failed
[2012-06-21 00:14:57] VERBOSE[28312] app_macro.c: == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/102-0000004f' in macro 'outisbusy'
[2012-06-21 00:14:57] VERBOSE[28312] pbx.c: == Spawn extension (from-internal, 0849999329, 6) exited non-zero on 'SIP/102-0000004f'
[2012-06-21 00:14:57] VERBOSE[28312] pbx.c: -- Executing [[email protected]:1] Hangup("SIP/102-0000004f", "") in new stack
[2012-06-21 00:14:57] VERBOSE[28312] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/102-0000004f'
2

hi

3

Ok - I know you are new but this was too much.

You posted 500 lines of irrelevant log and did not even bother to format it to make it easier to read.

There is no format g729a it’s only g.729 and it’s a licensed technology. Did you purchase g.729 channel licenses from Digium?

The bottom line is your trunk is not setup right. This is the only line in your log that matters:

[2012-06-21 00:14:53] VERBOSE[28312] pbx.c: -- Executing [[email protected]:3] NoOp("SIP/102-0000004f", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to other trunks") in new stack

I trimmed your post so someone else would be willing to read it.

4

Thanks for helping me with trimming it, Like you said, I’m new to this, I have been sitting here reading basically all the threads I could fine regarding this, I have struggled to get my trunk registered and after a long and winding road I got it to register my trunk-peer and sip. I’m just not sure what I have to fix on the trunk if it’s not setup right, I just thought when I got all extensions to work that it is OK, that’s why I have asked the question, HELP?

If there is any other info needed please let me know I will do my best to get it posted.

Regarding the G729, My sip or VIOP provider are using this codec, I didn’t know that I need to buy a License for this.

Thanks again.

5

g.729 is an ultra high compression CODEC and is licensed technology.

I am sure your carrier supports other CODEC’s.

The g.729 is the entire problem. If you don’t have the CODEC installed the call won’t complete.

6

Ok Thanks, I will get that installed, and I have spoken to my provider this is the only one they use, I am using a Linksys SPA-2102 and it’s also using the G729 and works find without the PBX but as you said… I will get that license and do the installation and try it then, Thanks so much

7

Linksys (or cisco) license g.729 for the device. All phones/ata’s have g.729 support.

This is an open source program, it is impossible to pay royalties to intellectual property owners.

8

Hi again, I have talked to my provider and they said that they have installed it for me, but still no outbound, I’m struggling with this for more then a month now. as on my linksys SPA2102 I see that it is g729a and not g729, if I set it to allow=g729&ulaw&alaw I just don’t get audio to my linksys but then if I set it to allow=g729a&ulaw&alaw I get audio but in the log it says the following,

[2012-06-22 14:44:51] WARNING[3619] frame.c: Cannot allow unknown format ‘g729a’
[2012-06-22 14:44:51] WARNING[3619] chan_sip.c: Codec configuration errors found in line 148 : allow = g729a

All this but my incoming works very well like I said previously even the IVR.

I’m really fed up with this.

If there is something ells you want me to post just let me know what,were and how… “I’m just not that good with this stuff, hardware is more my thing”

9

You carrier would not install the g.729, you have to go to Digium’s site and purchase the channel licenses.

I have been doing this a long time and I have never seen a carrier that only accepts g.729, it is ludricious as it cuts them out of many systems that don’t support that format.

g729a is g729 annex a, that is called g729 in Asterisk. You can see this with a “core show translation”


maieast*CLI> core show translation
         Translation times between formats (in microseconds) for one second of data
          Source Format (Rows) Destination Format (Columns)

           g723   gsm  ulaw  alaw g726aal2 adpcm  slin lpc10  g729 speex  ilbc  g726  g722 siren7 siren14 slin16  g719 speex16 testlaw
     g723     -     -     -     -        -     -     -     -     -     -     -     -     -      -       -      -     -       -       -
      gsm     -     -  1000  1000     5998  1998   999  5998 10998     -     -  5999  1999      -       -   4998     -       -    1000
     ulaw     -  4001     -     1     5000  1000     1  5000 10000     -     -  5001  1001      -       -   4000     -       -       2
     alaw     -  4001     1     -     5000  1000     1  5000 10000     -     -  5001  1001      -       -   4000     -       -       2
 g726aal2     -  6999  3000  3000        -  3998  2999  7998 12998     -     -  7999  3999      -       -   6998     -       -    3000
    adpcm     -  5000  1001  1001     5999     -  1000  5999 10999     -     -  6000  2000      -       -   4999     -       -    1001
     slin     -  4000     1     1     4999   999     -  4999  9999     -     -  5000  1000      -       -   3999     -       -       1
    lpc10     -  7999  4000  4000     8998  4998  3999     - 13998     -     -  8999  4999      -       -   7998     -       -    4000
     g729     -  6999  3000  3000     7998  3998  2999  7998     -     -     -  7999  3999      -       -   6998     -       -    3000
    speex     -     -     -     -        -     -     -     -     -     -     -     -     -      -       -      -     -       -       -
     ilbc     -     -     -     -        -     -     -     -     -     -     -     -     -      -       -      -     -       -       -
     g726     -  6999  3000  3000     7998  3998  2999  7998 12998     -     -     -  3999      -       -   6998     -       -    3000
     g722     -  6999  3000  3000     7998  3998  2999  7998 12998     -     -  7999     -      -       -   2999     -       -    3000
   siren7     -     -     -     -        -     -     -     -     -     -     -     -     -      -       -      -     -       -       -
  siren14     -     -     -     -        -     -     -     -     -     -     -     -     -      -       -      -     -       -       -
   slin16     - 10998  6999  6999    11997  7997  6998 11997 16997     -     - 11998  3999      -       -      -     -       -    6999
     g719     -     -     -     -        -     -     -     -     -     -     -     -     -      -       -      -     -       -       -
  speex16     -     -     -     -        -     -     -     -     -     -     -     -     -      -       -      -     -       -       -
  testlaw     -  4001     2     2     5000  1000     1  5000 10000     -     -  5001  1001      -       -   4000     -       -       -
maieast*CLI>

Notice the output from this system has numeric delay values next to g729, yours will have “-” indicating the CODEC is not installed.

If you get frustrated you will never accomplish your goals.

You can also end your frustration and support the project in one easy step, click the support link on the top of this page and purchase a support package. I would be glad to get you on track.

Scott

10

Hi,

First of all, I have to thank you for your support, I’m not that frustrated it’s a matter of speaking… but all is good I have made a screen capture of my “core show translation”

You might tell me if it’s installed or not, to me it looks the same as yours,I have paid my provider to load it on my FreePBX and if it’s not I’ll let them know,

I had to ask my provider to give me access to install my PBX, They have created my account and gave me access for my sip to register.

I have tried all types of dial plans even “X.” as I saw another guy is using saying it allows all numbers to go thru, every where I read the problem regarding “All circuits are busy now” points me in the direction of the dial plan.

If I could purchase a support package I would but at this time I’m a starting business 4 months running and just can’t afford it at this time, maybe in the near future if all is going well.

Thanks again for your trouble.

11

Never heard of a provider working on a system for the customer, that’s nice.

Look’s like it is installed. Good stuff.

If you look at the log just before the rejected you should see sip/xxx/yourdialednumber

Is the dialed number correct?

12

Ok, first of all, My provider is also my Head Office, We do wireless internet via mikrotik CPE’s and also VOIP services, I have started a reseller branch in my area so that’s why I get the good stuff. They wanted to install the complete package for me but I said NO I need to learn and get back into this kind of stuff…

Anyway I’ve got something right, I have change my outbound CID to my VIOP number and got it to dial out, but now I can dial my Head office ON NET numbers 7 digits and also our local phone company’s time line and directory inquiry with 4 digits but our normal land lines and Cell phone numbers with 10 digits give me a busy tone, I’m starting to wonder if they have opend that lines for me to dial… or maybe there is still something wrong.

All number I dial are correct on the log eg. 0849999329 the log shows the same number.

13

My Country is South Africa and we have a few dialing codes, just a few 011 Johannesburg, 012 Pretoria, 021 Capetown or 016 Gauteng this is just a few then we have 3 cell phone conpany’s 082, 083, 084 and then our VOIP lines 087 all of these are 10digit phone nunbers 012#######.

I will ask them and see what they say.

Thanks again

14

I should have added this, the top one is for my customer care line, it’s only 7 digits this is to my provider no need to dial the area code, the second one are for our local land line 4 digit directory inquiry. the top two are working no problem I can call my support guy’s as 2701040 and the 4 digit is working as well.

With the 10 digit I receive this on the log.
[2012-06-23 17:32:06] VERBOSE[28948] app_dial.c: – Called SIP/Screamer/849999329
[2012-06-23 17:32:07] VERBOSE[3596] chan_sip.c: – Got SIP response 603 “Declined” back from XX.XX.XX.XX:5060

I don’t know why would it decline my call.

15

Ok I have spoken to my provider, they need to see the number from dialed 0849999329 to 849999329 without the 0, I have used this outbound dial plan.

But still get the busy tone, are this dial plan right.

Thx

16

Only the bottom one is right. You have to prepend the common digits in scenario 1 and 2 to make the full 9 digits.

17

Ok, you need to ask your provider how they want it formatted. May be expecting entire e.164 string.

What Country/City is this?

18

I need to know how I can check for how many channels I have on the g.729 license.

19

Hi Scott,

Thanks for all your help, my pbx is working 100% now and can dial all numbers needed to be called, I have had to add the 0027 on at the prepend at the trunk dial rules and the 0 on the prefix at the outbound dial pattern. It is setup like the pic captures.

OUTBOUND DIAL

TRUNKS DIAL

Thanks for a great program.
now I just have to get my queues right.

Jaco

20

I am not sure if I am allowed to do this but this is the output from my asterisk cli:

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
– Executing [[email protected]:1] Macro(“SIP/6358-00000026”, “user-callerid,SKIPTTL,”) in new stack
– Executing [[email protected]:1] Set(“SIP/6358-00000026”, “AMPUSER=6358”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/6358-00000026”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/6358-00000026”, “1?Set(REALCALLERIDNUM=6358)”) in new stack
– Executing [[email protected]:4] Set(“SIP/6358-00000026”, “AMPUSER=6358”) in new stack
– Executing [[email protected]:5] Set(“SIP/6358-00000026”, “AMPUSERCIDNAME=Anushree Takalkar”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/6358-00000026”, “0?report”) in new stack
– Executing [[email protected]:7] Set(“SIP/6358-00000026”, “AMPUSERCID=6358”) in new stack
– Executing [[email protected]:8] Set(“SIP/6358-00000026”, “CALLERID(all)=“Anushree Takalkar” <6358>”) in new stack
– Executing [[email protected]:9] ExecIf(“SIP/6358-00000026”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [[email protected]:10] GotoIf(“SIP/6358-00000026”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [[email protected]:19] NoOp(“SIP/6358-00000026”, “Using CallerID “Anushree Takalkar” <6358>”) in new stack
– Executing [[email protected]:2] Set(“SIP/6358-00000026”, “_NODEST=”) in new stack
– Executing [[email protected]:3] Macro(“SIP/6358-00000026”, “record-enable,6358,OUT,”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/6358-00000026”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] AGI(“SIP/6358-00000026”, “recordingcheck,20130115-133002,1358274602.51”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck,20130115-133002,1358274602.51: Outbound recording not enabled
– <SIP/6358-00000026>AGI Script recordingcheck completed, returning 0
– Executing [[email protected]:5] MacroExit(“SIP/6358-00000026”, “”) in new stack
– Executing [[email protected]:4] Macro(“SIP/6358-00000026”, “dialout-trunk,3,0017408172415,86245,”) in new stack
– Executing [[email protected]:1] Set(“SIP/6358-00000026”, “DIAL_TRUNK=3”) in new stack
– Executing [[email protected]:2] GosubIf(“SIP/6358-00000026”, “1?sub-pincheck,s,1”) in new stack
– Executing [[email protected]:1] Authenticate(“SIP/6358-00000026”, “86245,”) in new stack
– <SIP/6358-00000026> Playing ‘agent-pass.gsm’ (language ‘en’)
– <SIP/6358-00000026> Playing ‘auth-thankyou.gsm’ (language ‘en’)
– Executing [[email protected]:2] ResetCDR(“SIP/6358-00000026”, “”) in new stack
– Executing [[email protected]:3] Return(“SIP/6358-00000026”, “”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/6358-00000026”, “0?disabletrunk,1”) in new stack
– Executing [[email protected]:4] Set(“SIP/6358-00000026”, “DIAL_NUMBER=0017408172415”) in new stack
– Executing [[email protected]:5] Set(“SIP/6358-00000026”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [[email protected]:6] Set(“SIP/6358-00000026”, “OUTBOUND_GROUP=OUT_3”) in new stack
– Executing [[email protected]:7] GotoIf(“SIP/6358-00000026”, “0?nomax”) in new stack
– Executing [[email protected]:8] GotoIf(“SIP/6358-00000026”, “0?chanfull”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/6358-00000026”, “0?skipoutcid”) in new stack
– Executing [[email protected]:10] Set(“SIP/6358-00000026”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [[email protected]:11] Macro(“SIP/6358-00000026”, “outbound-callerid,3”) in new stack
– Executing [[email protected]:1] ExecIf(“SIP/6358-00000026”, “0?Set(CALLERPRES()=)”) in new stack
– Executing [[email protected]:2] ExecIf(“SIP/6358-00000026”, “0?Set(REALCALLERIDNUM=6358)”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/6358-00000026”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [[email protected]:6] Set(“SIP/6358-00000026”, "USEROUTCID=“Anushree Takalkar " <2267803358>”) in new stack
– Executing [[email protected]:7] Set(“SIP/6358-00000026”, "EMERGENCYCID=“Anushree Takalkar " <2267803358>”) in new stack
– Executing [[email protected]:8] Set(“SIP/6358-00000026”, “TRUNKOUTCID=”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/6358-00000026”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [[email protected]:12] ExecIf(“SIP/6358-00000026”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [[email protected]:13] ExecIf(“SIP/6358-00000026”, "1?Set(CALLERID(all)=“Anushree Takalkar " <2267803358>)”) in new stack
– Executing [[email protected]:14] ExecIf(“SIP/6358-00000026”, “0?Set(CALLERPRES()=prohib_passed_screen)”) in new stack
– Executing [[email protected]:12] ExecIf(“SIP/6358-00000026”, “1?AGI(fixlocalprefix)”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
> fixlocalprefix: Using pattern 0119122|67803333
> fixlocalprefix: Using pattern 9122|67803333
> fixlocalprefix: Using pattern 001+614NXXXXXX
> fixlocalprefix: Using pattern 00+1614NXXXXXX
> fixlocalprefix: Using pattern 01191|22678032XX
– <SIP/6358-00000026>AGI Script fixlocalprefix completed, returning 0
– Executing [[email protected]:13] Set(“SIP/6358-00000026”, “OUTNUM=0017408172415”) in new stack
– Executing [[email protected]:14] Set(“SIP/6358-00000026”, “custom=IAX2/wdindiapri”) in new stack
– Executing [[email protected]:15] ExecIf(“SIP/6358-00000026”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))”) in new stack
– Executing [[email protected]:16] Macro(“SIP/6358-00000026”, “dialout-trunk-predial-hook,”) in new stack
– Executing [[email protected]:1] MacroExit(“SIP/6358-00000026”, “”) in new stack
– Executing [[email protected]:17] GotoIf(“SIP/6358-00000026”, “0?bypass,1”) in new stack
– Executing [[email protected]:18] GotoIf(“SIP/6358-00000026”, “0?customtrunk”) in new stack
– Executing [[email protected]:19] Dial(“SIP/6358-00000026”, “IAX2/wdindiapri/0017408172415,300,”) in new stack
– Called wdindiapri/0017408172415
– Call accepted by 192.168.40.50 (format ulaw)
– Format for call is ulaw
– Hungup ‘IAX2/wdindiapri-16733’
– No one is available to answer at this time (1:0/0/0)
– Executing [[email protected]:20] Goto(“SIP/6358-00000026”, “s-NOANSWER,1”) in new stack
– Goto (macro-dialout-trunk,s-NOANSWER,1)
– Executing [[email protected]:1] NoOp(“SIP/6358-00000026”, “Dial failed due to trunk reporting NOANSWER - giving up”) in new stack
– Executing [[email protected]:2] PlayTones(“SIP/6358-00000026”, “congestion”) in new stack
– Executing [[email protected]:3] Congestion(“SIP/6358-00000026”, “20”) in new stack
== Spawn extension (macro-dialout-trunk, s-NOANSWER, 3) exited non-zero on ‘SIP/6358-00000026’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 0017408172415, 4) exited non-zero on ‘SIP/6358-00000026’
– Executing [[email protected]:1] Macro(“SIP/6358-00000026”, “hangupcall”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/6358-00000026”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [[email protected]:4] GotoIf(“SIP/6358-00000026”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [[email protected]:7] GotoIf(“SIP/6358-00000026”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] Hangup(“SIP/6358-00000026”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/6358-00000026’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/6358-00000026’