Re: [asterisk-users] chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response
On 1/3/24 04:53, Henning Follmann wrote:
On Jan 2, 2024, at 23:17, the...@sys-concept.com wrote:
On 1/2/24 15:13, aster...@phreaknet.org wrote:
On 1/2/2024 4:03 PM, the...@sys-concept.com wrote:
I'm using asterisk-16.30.1
When I try to call another asterisk server over IAX I get a busy signal,
chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response
-- IAX2/192.168.143.1:4569-656 is circuit-busy
Asterisk-16.16 is working normally, no congestion error.
There is not enough information for anyone to really help or comment on this.
Dialplan and IAX2 configuration on both sides of the trunk?
CLI output on both sides with iax2 debug enabled?
It is very simple:
Local Asterisk, iax.conf:
[clinic_server]
type=friend
host=dynamic
context=internal
disallow=all
allow=ulaw
allow=alaw
requirecalltoken=no
callgroup=1
pickupgroup=1
extension.conf:
exten => 4,1,Dial(IAX2/home_server:@${clinic_server}/${EXTEN},60,rw)
exten => 4,n,Hangup() Remote Asterisk iax.conf:
[home_server]
type=friend
host=dynamic
secret=
context=extensions
disallow=all
allow=ulaw
allow=alaw
callgroup=1
pickupgroup=1
Remote extension.conf:
exten => 4,1,Dial(SIP/4,15,trw)
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?vmail:line2)
exten => 4,n(line2),Dial(SIP/54,20,rw)
exten => 4,n(vmail),Voicemail(4)
exten => 4,n,Voicemail(4)
exten => 4,n,Hangup()
You have no internal context in your dialplan. But in your iax.conf you specify
internal as your context.
-H
I forgot to write, the remote asterisk has:
[internal]
...
include => extensions
[extensions]
exten => 4,1,Dial(SIP/4,15,trw)
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?vmail:line2)
exten => 4,n(line2),Dial(SIP/54,20,rw)
exten => 4,n(vmail),Voicemail(4)
exten => 4,n,Voicemail(4)
exten => 4,n,Hangup()
So on the remote asterisk there is context [internal]
I even noticed starting asterisk-16.30 is much slower than starting
Asterisk-16.16
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Re: [asterisk-users] chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response
On 1/2/24 15:13, aster...@phreaknet.org wrote:
On 1/2/2024 4:03 PM, the...@sys-concept.com wrote:
I'm using asterisk-16.30.1
When I try to call another asterisk server over IAX I get a busy signal,
chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response
-- IAX2/192.168.143.1:4569-656 is circuit-busy
Asterisk-16.16 is working normally, no congestion error.
There is not enough information for anyone to really help or comment on this.
Dialplan and IAX2 configuration on both sides of the trunk?
CLI output on both sides with iax2 debug enabled?
It is very simple:
Local Asterisk, iax.conf:
[clinic_server]
type=friend
host=dynamic
context=internal
disallow=all
allow=ulaw
allow=alaw
requirecalltoken=no
callgroup=1
pickupgroup=1
extension.conf:
exten => 4,1,Dial(IAX2/home_server:@${clinic_server}/${EXTEN},60,rw)
exten => 4,n,Hangup() Remote Asterisk iax.conf:
[home_server]
type=friend
host=dynamic
secret=
context=extensions
disallow=all
allow=ulaw
allow=alaw
callgroup=1
pickupgroup=1
Remote extension.conf:
exten => 4,1,Dial(SIP/4,15,trw)
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?vmail:line2)
exten => 4,n(line2),Dial(SIP/54,20,rw)
exten => 4,n(vmail),Voicemail(4)
exten => 4,n,Voicemail(4)
exten => 4,n,Hangup()
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[asterisk-users] chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response
I'm using asterisk-16.30.1 When I try to call another asterisk server over IAX I get a busy signal, chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response -- IAX2/192.168.143.1:4569-656 is circuit-busy Asterisk-16.16 is working normally, no congestion error. -- Thelma -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium or Sangoma? What happened to Digium cards
I'm disappointed with Sangoma! I have one of those Digium S101i (iaxy) adapters that is still working (in production), doesn't need any drivers. After, I purchased Sangoma USBfxo (U100) adapter, never had a chance to use it (still brand new in a box) that require some extra driver to run it, driver that is no longer available. And the U100 is obsolete. So my advise stay away from Sangoma Joseph On 1/12/21 4:17 PM, John Kiniston wrote: > Sangoma purchased Digium. > > You can find Sangoma cards at https://www.sangoma.com/telephony-cards/ > > On Tue, Jan 12, 2021 at 2:29 PM bilal ghayyad wrote: > >> Hello All; >> >> We were using Digium cards, now I am not able to reach for digium website >> that contains the telephony cards and Asterisk website currently is taking >> us for Sangoma, so what happened in Digium cards? >> >> Regards >> Bilal >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is not available.
On 1/4/21 12:01 PM, Doug Lytle wrote:
Gosub(check-number-forwarding,Dial(SIP/718xxpstn-5665,20,U(atb-sub))
>
Is ARG1 = atb-sub ?
>
> No.
>
> My complete line
>
> exten => _45XX,1,Set(_ARG1=${EXTEN}
> same => n,Gosub(check-number-forwarding,s,1(${ARG1}))
>
> So, if someone were to dial a 4 digit number starting with 45 (i.e. 4522), it
> would jump to the sub-routine called check-number-forwarding and supply the
> variable of 4522 to that sub-routine.
>
> It could have been just as easily written as
>
> same => n,Gosub(check-number-forwarding,s,1(4522))
>
> Your sub-routine will need to pass what dialing options you are wanting to
> use.
>
> A good source of information
>
> https://wiki.asterisk.org/wiki/display/AST/Gosub
>
> Doug
OK, both combination worked but still silence until the all numbers are dialed.
exten => 65,1,Progress() ;plays the music on internal call
exten => 65,n,Answer
exten => 65,n,Dial(SIP/718xx@pstn-5665,20,U(atb-sub))
[atb-sub]
exten => s,1,Wait(6)
exten => s,n,SendDTMF(1)
exten => s,n,SendDTMF(x#)
exten => s,n,SendDTMF(x#)
exten => s,n,SendDTMF(1)
[atb-sub]
exten => s,1,Wait(6)
exten => s,n,SendDTMF(1)
exten => s,n,Gosub(check-number-forwarding,s,1(xx#))
exten => s,n,Gosub(check-number-forwarding,s,1(#)
exten => s,n,SendDTMF(1)
What I want is hear the phone dialing the number or play music, so I know it is
dialing or doing something.
in the first line I have "Progress()" it used to play music but it doesn't
anymore.
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Re: [asterisk-users] ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is not available.
On 1/4/21 10:44 AM, Doug Lytle wrote:
>
How do you enable the phone speaker on the Gosub?
>
I had:
Dial(SIP/718x@pstn-5665,20,m(default)M(atb))
>
> You can provide variables to your gosub routine, for an example
>
> Gosub(check-number-forwarding,s,1(${ARG1}))
>
> Doug
This one work (but no dialing numbers on phone speeker)
Dial(SIP/718x@pstn-5665,20,U(atb-sub))
This one doesn't work:
Gosub(check-number-forwarding,Dial(SIP/718xxpstn-5665,20,U(atb-sub))
Is ARG1 = atb-sub ?
Thanks for your help.
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Re: [asterisk-users] ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is not available.
On 1/4/21 9:04 AM, Joshua C. Colp wrote: > On Mon, Jan 4, 2021 at 11:57 AM wrote: > >> Did execution of macro changed in Astersik-16.15 ? >> >> When I try to dial an extension that call macro I get an error: >> >> app.c:280 ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is >> not available. >> >> Dial(SIP/718xx@pstn-5665,20,m(default)M(atb)) >> > > The Macro application is not built by default, it has to be explicitly > selected in "make menuselect" when building Asterisk. Have you done so? On Gentoo we build Asterisk with emerge and use "flags" to enable or disable certain features: Installed versions: 16.15.1-r1 (alsa bluetooth caps iconv ssl vorbis -calendar -cluster -curl -dahdi -debug -doc -freetds -gtalk -http -ilbc -ldap -libressl -lua -mysql -newt -odbc -oss -pjproject -portaudio -postgres -radius -selinux -snmp -span -speex -srtp -static -statsd -syslog -unbound -xmpp Which one of the above (the one with "-" in front) are responsible for compiling in Macro feature. I an replace macro with Gosub but my phone doesn't play the dialing tones after I did it. My macro execute several sequence of numbers after connecting and I can hear them in the speakerphone now with Gosub, it does execute these numbers but I don't hear anything in the speakerphone just silence. Is there a way to activate speakerphone what on Gosub? So I know the phone is dialing the numbers? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is not available.
On 1/4/21 10:09 AM, Doug Lytle wrote: app.c:280 ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is not available. > > Macros are no longer built by default in Asterisk 16. This was documented in > the UPGRADE.txt file > > app_macro: > - The app_macro module is now deprecated and by default it is no longer >built. Users should migrate to app_stack (Gosub). A warning is logged >the first time any Macro is used. > > Doug Thanks, I was able to figure it out, replace the subroutine witht Gosub How do you enable the phone speaker on the Gosub? I had: Dial(SIP/718x@pstn-5665,20,m(default)M(atb)) replace it with: Dial(SIP/718xx@pstn-5665,20,U(atb-sub)) My macro used to execute sequence of numbers after it connects, when I replace the macro with Gosub I don't hear phone dialing the numbers on the phone that macro used to execute (just silence). Is there a way to play these numbers on the phone (so I know it is working)? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is not available.
Did execution of macro changed in Astersik-16.15 ? When I try to dial an extension that call macro I get an error: app.c:280 ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is not available. Dial(SIP/718xx@pstn-5665,20,m(default)M(atb)) -- Thelma -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 13 takes over an hour to clear the MWI light
I just tired/upgraded to asterisk-16.13.0 and the problem still persist. Looking at some post on digium.support web-page, they don't have a clear solution or know what causing it. On 12/24/2020 05:03 AM, Julian Beach wrote: > Hello Thelma, > > Thursday, December 24, 2020, 9:26:53 AM, the...@sys-concept.com wrote: > >> In astersik-11 MWI light was cleared as soon as I checked the message. >> In asterink-13 it takes about 20min to set the light ON and the light >> takes over an hour to clear. > > I had this problem following an upgrade between releases of Asterisk 13 last > year, but I upgraded to Asterisk 16 and the problem went away without any > need for configuration changes. > > Julian > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 13 takes over an hour to clear the MWI light
In astersik-11 MWI light was cleared as soon as I checked the message. In asterink-13 it takes about 20min to set the light ON and the light takes over an hour to clear. What had changed? In sip.cong [400] ... mailbox=400 voicemail.conf [default] 400 => ,user, email -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk-13 MWI - phone not blinking
On asterisk-11 MWI was working correctly, phone message light was blinking (standard phone). With asterisk-13, this feature is not working. Who to trouble shoot? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] db_execute_sql: Error executing SQL (COMMIT): database is locked
In: /var/lib/asterisk -rw-r--r-- 1 asterisk asterisk 12288 Dec 23 10:52 astdb.sqlite3 ast_db_put: Couldn't execute statement: SQL logic error ast_db_put: Couldn't execute statement: attempt to write a readonly database db_execute_sql: Error executing SQL (COMMIT): database is locked The astdb.sqlite3 was root:root but I change it to asterisk:asterisk and I still getting these error messages -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk Unknown DYNAMIC_FEATURES item 'automon' on channel
On 12/23/2020 09:54 AM, Doug Lytle wrote: > Review your features.conf file in /etc/asterisk > > Doug I found id. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk Unknown DYNAMIC_FEATURES item 'automon' on channel
I just upgraded to asterisk-13 (from 11) and I get some errors: 1.) Unknown DYNAMIC_FEATURES item 'automon' on channel SIP Unknown DYNAMIC_FEATURES item 'automon' on channel IAX2/voip Does anybody know how to get rid of them? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] upgrade asterisk 11 to 13 or 11-16
Sound reasonable. I know it take time to debug the dial-plan after upgrade. Can I use sipp, from command line to call my local asterisk specific extension and to observe in another terminal via "asterisk -vvr" what it is doing? On 12/07/2020 11:50 AM, Eric Wieling wrote: > Read UPGRADE.TXT in v13 and v16. Then read it again. > > I upgraded from Asterisk v11 to Asterisk v13. Once all issues were > resolved, then I switched to PJSIP. Once all the issues with PJSIP > were resolved, then I upgraded from v13 to Asterisk v16. This was done > over the course of about a year, but I was not in any hurry. > > PJSIP configuration is fundamentally different chan_sip configuration. I > don't recommend switching to PJSIP and upgrade Asterisk at the same time. > > On 12/6/20 3:38 PM, the...@sys-concept.com wrote: >> I'm planning to upgrade my asterisk-11.25 to ver. 13 >> or should I go to 11 to 16 >> >> Is there any official documentation how to upgrade, what to watch for >> during upgrade? >> >> > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] upgrade asterisk 11 to 13 or 11-16
On 12/07/2020 05:06 AM, Jöran Vinzens wrote: > Hi, > > I guess describing how SIPp works here on a mailliste might be too much. > But if you do not want to prove your setup automatically, you do not need > to know SIPp. > > But there was a talk in 2014 Astricon giving an overview about SIP Testing > with SIPp > https://www.youtube.com/watch?v=TZMrPJM4HMc > > BR > Jöran > > > On Sun, Dec 6, 2020 at 11:25 PM wrote: > >> On 12/06/2020 01:44 PM, Jöran Vinzens wrote: >>> Hi, >>> >>> I did a talk on Astricon 2019 on this topic. Unfortunately there are no >>> videos of that year but you can find my slides here covering some >> pitfalls. >>> >> https://www.slideshare.net/JranVinzens/asterisk-11to16-what-could-go-wrong >>> >>> Good luck by updating. >>> >>> BR >>> Jöran >>> >>> >>> schrieb am So., 6. Dez. 2020, 21:40: >>> I'm planning to upgrade my asterisk-11.25 to ver. 13 or should I go to 11 to 16 Is there any official documentation how to upgrade, what to watch for during upgrade? >> Thanks for the input. I've never run SIPp signaling test. >> Is there more information how to implement it? Thanks, yes initially it looked interesting. But I don't see how "sipp" can be use to test my extension.conf dial plan. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] upgrade asterisk 11 to 13 or 11-16
On 12/06/2020 01:44 PM, Jöran Vinzens wrote: > Hi, > > I did a talk on Astricon 2019 on this topic. Unfortunately there are no > videos of that year but you can find my slides here covering some pitfalls. > https://www.slideshare.net/JranVinzens/asterisk-11to16-what-could-go-wrong > > Good luck by updating. > > BR > Jöran > > > schrieb am So., 6. Dez. 2020, 21:40: > >> I'm planning to upgrade my asterisk-11.25 to ver. 13 >> or should I go to 11 to 16 >> >> Is there any official documentation how to upgrade, what to watch for >> during upgrade? >> Thanks for the input. I've never run SIPp signaling test. Is there more information how to implement it? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] upgrade asterisk 11 to 13 or 11-16
I'm planning to upgrade my asterisk-11.25 to ver. 13 or should I go to 11 to 16 Is there any official documentation how to upgrade, what to watch for during upgrade? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [SOLVED] incoming call label
On 02/16/2018 12:27 AM, Jean Aunis wrote: > Le 16/02/2018 à 05:30, the...@sys-concept.com a écrit : >> On 02/15/2018 04:49 PM, Joshua Colp wrote: >>> On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: >>> >>> >>> >>>> Thanks again for the hint. >>>> Here is the output from asterisk. >>>> >>>> The call is coming on Audocodes gateway from: pstn- >>>> >>>> But asterisk display: >>>> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 >>>> >>>> Why not loolking up "pstn-" in sip.conf? >>> It found pstn- using 10.10.0.8:5060 - if the request always comes >>> from the same IP address and port it has no other way built in to >>> differentiate between the two except by matching based on username in >>> the 'From' header. >> It didn't find "pstn- using 10.10.0.8:5060" >> The call came IN from PSTN line on audiocodes equipment to FXO port that >> is labelled "pstn-" so asterisk reported as such. >> And I think asterisk suppose to lookup this label in sip.conf to the >> registered entry but instead selected pstn-9998 entry; I don't know why. >> >> If the call came IN on pstn- >> and sip.conf has two entries: >> [pstn-] >> [pstn-9998] >> >> Why it can not distinguish between the two of them correctly? >> >> -- >> Thelma >> >> > If your device supports SIP authentication, you can try to turn on the > "match_auth_username" parameter in sip.conf. It is said to be > experimental but has always worked well for me. Thanks for the input. I've tried "match_auth_username" parameter in sip.conf it didn't work. but the above entry needs to be enable in sip.conf to avoid "user mismatch" error Calls are coming on "pstn-" from Audiocodes FXO but sip.conf recognizes it as "pstn-9998" But adding to sip.conf relevant entry in my case [pstn-] and [pstn-9998] "user=peer" SOLVED the problem. Now command line is showing correctly: calls coming from pstn- are showing on command line as "pstn-" Thanks to Julian Beach for the hint! -- Thelma. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [SOLVED] username mismatch
On 02/16/2018 10:50 AM, the...@sys-concept.com wrote: > When I have an incoming call I get a "username mismatch": > > WARNING[7459][C-0007]: chan_sip.c:16587 check_auth: username mismatch, > have <55>, digest has > NOTICE[7459][C-0007]: chan_sip.c:25809 handle_request_invite: Failed to > authenticate device "KZ" <sip:7804715665@10.0.0.110>;tag=1c117168544 > > my sip.conf entry: > > [pstn-1270] ; incoming/outoging calls on FXO > type=friend > secret=x > username=voice-1270 > user=peer > mailbox=369 ; just for audiocodes error complain > host=dynamic > insecure=port,invite > canreinvite=no ; (dtmf not working correctly without this one) > disallow=all > allow=gsm > allow=ulaw > allow=alaw > nat=no > context=incoming > callgroup=1 > pickupgroup=1 > qualify=yes > sip.conf needs to have enabled: match_auth_username=yes -- Thelma -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] username mismatch
When I have an incoming call I get a "username mismatch": WARNING[7459][C-0007]: chan_sip.c:16587 check_auth: username mismatch, have <55>, digest has NOTICE[7459][C-0007]: chan_sip.c:25809 handle_request_invite: Failed to authenticate device "KZ" <sip:7804715665@10.0.0.110>;tag=1c117168544 my sip.conf entry: [pstn-1270] ; incoming/outoging calls on FXO type=friend secret=x username=voice-1270 user=peer mailbox=369 ; just for audiocodes error complain host=dynamic insecure=port,invite canreinvite=no ; (dtmf not working correctly without this one) disallow=all allow=gsm allow=ulaw allow=alaw nat=no context=incoming callgroup=1 pickupgroup=1 qualify=yes -- Thelma -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
On 02/15/2018 04:49 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: > > > >> >> Thanks again for the hint. >> Here is the output from asterisk. >> >> The call is coming on Audocodes gateway from: pstn- >> >> But asterisk display: >> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 >> >> Why not loolking up "pstn-" in sip.conf? > > It found pstn- using 10.10.0.8:5060 - if the request always comes from > the same IP address and port it has no other way built in to differentiate > between the two except by matching based on username in the 'From' header. It didn't find "pstn- using 10.10.0.8:5060" The call came IN from PSTN line on audiocodes equipment to FXO port that is labelled "pstn-" so asterisk reported as such. And I think asterisk suppose to lookup this label in sip.conf to the registered entry but instead selected pstn-9998 entry; I don't know why. If the call came IN on pstn- and sip.conf has two entries: [pstn-] [pstn-9998] Why it can not distinguish between the two of them correctly? -- Thelma -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
Thelma On 02/15/2018 07:16 PM, the...@sys-concept.com wrote: > > On 02/15/2018 04:49 PM, Joshua Colp wrote: >> On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: >> >> >> >>> >>> Thanks again for the hint. >>> Here is the output from asterisk. >>> >>> The call is coming on Audocodes gateway from: pstn- >>> >>> But asterisk display: >>> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 >>> >>> Why not loolking up "pstn-" in sip.conf? >> >> It found pstn- using 10.10.0.8:5060 - if the request always comes from >> the same IP address and port it has no other way built in to differentiate >> between the two except by matching based on username in the 'From' header. >> > > Call comes from same IP address always. > To comes form Audiocode: > > <--- SIP read from UDP:10.10.0.8:5060 ---> > INVITE sip:4@10.10.0.4 SIP/2.0 > Via: SIP/2.0/UDP 10.10.0.8;branch=z9hG4bKac766808844 > Max-Forwards: 70 > From: "Z" <sip:7804715665@10.10.0.8>;tag=1c766802762 > To: <sip:4@10.10.0.4> > Call-ID: 7668022781522018162620@10.10.0.8 > CSeq: 1 INVITE > Contact: <sip:pstn-@10.10.0.8:5060> > > Contact: "sip:pstn-" > > And it found in sip.conf only: > Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 > > Is perhaps the name effected by the special character "-" (dash) that is > why it only matches "pstn" and take the first one it found. Will it > make a difference if I rename the port to pstn_ in configuration files. > > -- > Thelma sip show peers Name/username HostDyn Forcerport ComediaACL Port Status Description pstn-/voice- 10.10.0.8D No No 5060 Unmonitored pstn-9998/fax-999810.10.0.8D No No 5060 Unmonitored -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
On 02/15/2018 04:49 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote: > > > >> >> Thanks again for the hint. >> Here is the output from asterisk. >> >> The call is coming on Audocodes gateway from: pstn- >> >> But asterisk display: >> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 >> >> Why not loolking up "pstn-" in sip.conf? > > It found pstn- using 10.10.0.8:5060 - if the request always comes from > the same IP address and port it has no other way built in to differentiate > between the two except by matching based on username in the 'From' header. > Call comes from same IP address always. To comes form Audiocode: <--- SIP read from UDP:10.10.0.8:5060 ---> INVITE sip:4@10.10.0.4 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.8;branch=z9hG4bKac766808844 Max-Forwards: 70 From: "Z" <sip:7804715665@10.10.0.8>;tag=1c766802762 To: <sip:4@10.10.0.4> Call-ID: 7668022781522018162620@10.10.0.8 CSeq: 1 INVITE Contact: <sip:pstn-@10.10.0.8:5060> Contact: "sip:pstn-" And it found in sip.conf only: Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 Is perhaps the name effected by the special character "-" (dash) that is why it only matches "pstn" and take the first one it found. Will it make a difference if I rename the port to pstn_ in configuration files. -- Thelma -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming call label
On 02/15/2018 04:08 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 7:03 PM, the...@sys-concept.com wrote:
>> On 02/15/2018 03:44 PM, Joshua Colp wrote:
>>> On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
>>>> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
>>>>
>>>> IN audocodes setting I have:
>>>> "EndPoint Phone Number"
>>>>
>>>> Channel: 3phone number: pstn-
>>>> Channel: 4phone number: pstn-9998
>>>>
>>>> When I am calling " pstn-" the port number "Channel:3" lights up but
>>>> asterisk is showing that the call is coming on "pstn-9998"
>>>>
>>>> -- Executing . Answer("SIP/pstn-9998
>>>>
>>>> Asterisk should be showing "pstn-" (not pstn-9998)
>>>> Where is this label coming from?
>>>
>>> It is from the SIP entry in sip.conf that it was matched against.
>>>
>>
>> Thanks for the input.
>>
>> In sip.conf I have relevant entries.
>>
>> [pstn-] ; incoming/outgoing calls on FXO port
>> type=friend
>> secret=spa354
>> username=voice-
>> mailbox=622 ; just for audiocodes error complain
>> host=dynamic
>> canreinvite=no ; (dtmf not wroking correctly without this one)
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> nat=no
>> context=incoming
>> callgroup=1
>> pickupgroup=1
>> insecure=invite
>>
>> [pstn-9998]
>> type=friend
>> secret=158567
>> username=fax-9998
>> insecure=invite
>> mailbox=622 ; just for audiocodes error complain
>> host=dynamic
>> canreinvite=no ; (dtmf not wroking correctly without this one)
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> nat=no
>> context=incoming
>> callgroup=1
>> pickupgroup=
>>
>> My asterisk registration is correct as well:
>> sip show users
>> Username Secret Accountcode Def.Context
>> ACL Forcerport
>> pstn-9998 158567 incoming
>> No No
>> pstn- spa354 incoming
>> No No
>>
>> Caller display ID from PSTN on FXO ports are working OK.
>> The [pstn-] is channel: 4
>> The [pstn-9998] is channel: 3
>>
>> If the call on Audocode is lighting UP "channel:3" the sip.conf should
>> associate that call with [pstn-] (and not [pstn-9998])
>
> Not necessarily. You appear to be doing IP+port based matching. If requests
> always come from the same source IP address and port, then it would match
> only one. Turning on sip debug using "sip set debug on" and verbosity using
> "core set debug 9" would give you more information about each packet
> (including where it is from) and what was actually matched based on it.
Thanks again for the hint.
Here is the output from asterisk.
The call is coming on Audocodes gateway from: pstn-
But asterisk display:
Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060
Why not loolking up "pstn-" in sip.conf?
<--- SIP read from UDP:10.10.0.8:5060 --->
INVITE sip:4@10.10.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.8;branch=z9hG4bKac766808844
Max-Forwards: 70
From: "Z" <sip:7804715665@10.10.0.8>;tag=1c766802762
To: <sip:4@10.10.0.4>
Call-ID: 7668022781522018162620@10.10.0.8
CSeq: 1 INVITE
Contact: <sip:pstn-@10.10.0.8:5060>
Supported:
em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 249
v=0
o=AudiocodesGW 766797875 766797759 IN IP4 10.10.0.8
s=Phone-Call
c=IN IP4 10.10.0.8
t=0 0
m=audio 6000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<->
--- (14 headers 12 lines) ---
Sending to 10.10.0.8:5060 (no NAT)
Sending to 10.10.0.8:5060 (no NAT)
Using INVITE request as basis request - 7668022781522018162620@10.10.0.8
Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer -
audio=(ulaw|alaw)/video=(nothing)/
Re: [asterisk-users] incoming call label
On 02/15/2018 03:44 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
>> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
>>
>> IN audocodes setting I have:
>> "EndPoint Phone Number"
>>
>> Channel: 3phone number: pstn-
>> Channel: 4phone number: pstn-9998
>>
>> When I am calling " pstn-" the port number "Channel:3" lights up but
>> asterisk is showing that the call is coming on "pstn-9998"
>>
>> -- Executing . Answer("SIP/pstn-9998
>>
>> Asterisk should be showing "pstn-" (not pstn-9998)
>> Where is this label coming from?
>
> It is from the SIP entry in sip.conf that it was matched against.
>
Thanks for the input.
In sip.conf I have relevant entries.
[pstn-] ; incoming/outgoing calls on FXO port
type=friend
secret=spa354
username=voice-
mailbox=622 ; just for audiocodes error complain
host=dynamic
canreinvite=no ; (dtmf not wroking correctly without this one)
disallow=all
allow=ulaw
allow=alaw
nat=no
context=incoming
callgroup=1
pickupgroup=1
insecure=invite
[pstn-9998]
type=friend
secret=158567
username=fax-9998
insecure=invite
mailbox=622 ; just for audiocodes error complain
host=dynamic
canreinvite=no ; (dtmf not wroking correctly without this one)
disallow=all
allow=ulaw
allow=alaw
nat=no
context=incoming
callgroup=1
pickupgroup=
My asterisk registration is correct as well:
sip show users
Username Secret Accountcode Def.Context
ACL Forcerport
pstn-9998 158567 incoming
No No
pstn- spa354 incoming
No No
Caller display ID from PSTN on FXO ports are working OK.
The [pstn-] is channel: 4
The [pstn-9998] is channel: 3
If the call on Audocode is lighting UP "channel:3" the sip.conf should
associate that call with [pstn-] (and not [pstn-9998])
--
Thelma
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[asterisk-users] incoming call label
I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
IN audocodes setting I have:
"EndPoint Phone Number"
Channel: 3phone number: pstn-
Channel: 4phone number: pstn-9998
When I am calling " pstn-" the port number "Channel:3" lights up but
asterisk is showing that the call is coming on "pstn-9998"
-- Executing . Answer("SIP/pstn-9998
Asterisk should be showing "pstn-" (not pstn-9998)
Where is this label coming from?
--
Thelma
--
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Re: [asterisk-users] IAX port 4569
yes it does. netstat -nap | grep 4569 udp0 0 0.0.0.0:45690.0.0.0:* 17375/asterisk Thelma On 06/05/2017 03:10 PM, Helvio Junior wrote: > Use the command bellow to check if is Asterisk opening the port. > > netstat -nap | grep 4569 > > You need to see something like this output, otherwise your asterisk is > not opening the port. > > udp0 0 0.0.0.0:4569 0.0.0.0:* > 10244/asterisk > > Att, > Hélvio Junior > dCAA - Digium Certified Asterisk Administrator > SafeId - Gestão de identidades e Acessos > +55 41 | 9 9855-9300, single-sign-on.com.br > helvio.jun...@safetrend.com.br > > Em 05/06/2017 17:22, the...@sys-concept.com escreveu: >> Yes, it is working! >> >> tcpdump -ni any port 4569 >> dropped privs to tcpdump >> tcpdump: verbose output suppressed, use -v or -vv for full protocol >> decode >> listening on any, link-type LINUX_SLL (Linux cooked), capture size >> 262144 bytes >> 14:20:42.184521 IP 10.0.0.108.4569 > 10.0.0.100.4569: UDP, length 53 >> 14:20:42.184921 IP 10.0.0.100.4569 > 10.0.0.108.4569: UDP, length 37 >> 14:20:42.190529 IP 10.0.0.108.4569 > 10.0.0.100.4569: UDP, length 83 >> 14:20:42.190639 IP 10.0.0.100.4569 > 10.0.0.108.4569: UDP, length 71 >> 14:20:42.191378 IP 10.0.0.108.4569 > 10.0.0.100.4569: UDP, length 12 >> 14:20:45.320191 IP 192.168.141.7.4569 > 192.168.141.1.4569: UDP, >> length 31 >> 14:20:45.338718 IP 192.168.141.1.4569 > 192.168.141.7.4569: UDP, >> length 65 >> 14:20:45.338875 IP 192.168.141.7.4569 > 192.168.141.1.4569: UDP, >> length 82 >> 14:20:45.357173 IP 192.168.141.1.4569 > 192.168.141.7.4569: UDP, >> length 40 >> 14:20:45.357331 IP 192.168.141.7.4569 > 192.168.141.1.4569: UDP, >> length 65 >> 14:20:45.376559 IP 192.168.141.1.4569 > 192.168.141.7.4569: UDP, >> length 53 >> 14:20:45.376630 IP 192.168.141.7.4569 > 192.168.141.1.4569: UDP, >> length 12 >> ^C >> 12 packets captured >> 12 packets received by filter >> 0 packets dropped by kernel >> >> >> Thelma >> On 06/05/2017 02:17 PM, Marcelo Terres wrote: >>> You can use tcpdump in your server to verify if it is receiving the >>> packets. >>> >>> tcpdump -ni any port 4569 >>> >>> So you have more than one ip in the server? >>> >>> On 5 Jun 2017 9:13 pm, <the...@sys-concept.com> wrote: >>> >>>> No, I don't think it is IP table issue, I've not upgraded dd-wrt for a >>>> while and it was zoiper was working OK with my previous version of >>>> asterisk. >>>> >>>> After upgrade to 11.25.1 it stop working. >>>> I'm sure port forwarding on dd-wrt is working OK as I have port 80 and >>>> 443 open. >>>> >>>> >>>> Thelma >>>> On 06/05/2017 07:12 AM, Christopher van de Sande wrote: >>>>> Another might be to make sure iptables isn't blocking the connection. >>>>> >>>>> You can run >>>>> iptables -L -n -v >>>>> To see if its set to block any ports. >>>>> >>>>> >>>>> On June 5, 2017 9:06:55 AM EDT, the...@sys-concept.com wrote: >>>>>> I'm getting: >>>>>> netstat -a |grep 4569 >>>>>> udp0 0 0.0.0.0:45690.0.0.0:* >>>>>> >>>>>> Should I be getting localhost IP? >>>>>> >>>>>> Thelma >>>>>> >>>>>> On 06/05/2017 06:48 AM, the...@sys-concept.com wrote: >>>>>>> Does asterisk listen on port 4569 by default? >>>>>>> >>>>>>> I'm running version Asterisk 11.25.1 and have a problem registering >>>>>>> Zoiper (IAX) to Asterisk. >>>>>>> I'm getting an error: >>>>>>> Registration refused >>>>>>> >>>>>> -- >>>>>> _ >>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>>>>> >>>>>> Check out the new Asterisk community forum at: >>>>>> https://community.asterisk.org/ >>>>>> >>>>>> New to Asterisk? Start here: >>>>>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >>>>>> >>>>>> asterisk-users mailing list >>>>>> To UNSUBSCRIBE or update options visit: >>>>>>http://lists.digium.com/mailman/listinfo/asterisk-users >>>>> >>>>> >>>> -- >>>> _ >>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>>> >>>> Check out the new Asterisk community forum at: >>>> https://community.asterisk. >>>> org/ >>>> >>>> New to Asterisk? Start here: >>>>https://wiki.asterisk.org/wiki/display/AST/Getting+Started >>>> >>>> asterisk-users mailing list >>>> To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>> >>> >>> > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *****SPAM***** Re: IAX port 4569
Doesn't matter how much I increase the verbose output asterisk -vvr asterisk will not even print a single line. How to find out if my firewall has this port open? https://www.grc.com is reporting that my port is 4569 is in Stealth mode (so it is closed) :-/ Thelma On 06/05/2017 02:19 PM, Victor Villarreal wrote: > I think you need to increase verbose output and search in > /var/log/asterisk/full for any error message related to IAX2 registration > or simil. > > 2017-06-05 17:12 GMT-03:00 <the...@sys-concept.com>: > >> No, I don't think it is IP table issue, I've not upgraded dd-wrt for a >> while and it was zoiper was working OK with my previous version of >> asterisk. >> >> After upgrade to 11.25.1 it stop working. >> I'm sure port forwarding on dd-wrt is working OK as I have port 80 and >> 443 open. >> >> >> Thelma >> On 06/05/2017 07:12 AM, Christopher van de Sande wrote: >>> Another might be to make sure iptables isn't blocking the connection. >>> >>> You can run >>> iptables -L -n -v >>> To see if its set to block any ports. >>> >>> >>> On June 5, 2017 9:06:55 AM EDT, the...@sys-concept.com wrote: >>>> I'm getting: >>>> netstat -a |grep 4569 >>>> udp0 0 0.0.0.0:45690.0.0.0:* >>>> >>>> Should I be getting localhost IP? >>>> >>>> Thelma >>>> >>>> On 06/05/2017 06:48 AM, the...@sys-concept.com wrote: >>>>> Does asterisk listen on port 4569 by default? >>>>> >>>>> I'm running version Asterisk 11.25.1 and have a problem registering >>>>> Zoiper (IAX) to Asterisk. >>>>> I'm getting an error: >>>>> Registration refused >>>>> >>>> >>>> -- >>>> _ >>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>>> >>>> Check out the new Asterisk community forum at: >>>> https://community.asterisk.org/ >>>> >>>> New to Asterisk? Start here: >>>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >>>> >>>> asterisk-users mailing list >>>> To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: https://community.asterisk. >> org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX port 4569
Yes, it is working! tcpdump -ni any port 4569 dropped privs to tcpdump tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on any, link-type LINUX_SLL (Linux cooked), capture size 262144 bytes 14:20:42.184521 IP 10.0.0.108.4569 > 10.0.0.100.4569: UDP, length 53 14:20:42.184921 IP 10.0.0.100.4569 > 10.0.0.108.4569: UDP, length 37 14:20:42.190529 IP 10.0.0.108.4569 > 10.0.0.100.4569: UDP, length 83 14:20:42.190639 IP 10.0.0.100.4569 > 10.0.0.108.4569: UDP, length 71 14:20:42.191378 IP 10.0.0.108.4569 > 10.0.0.100.4569: UDP, length 12 14:20:45.320191 IP 192.168.141.7.4569 > 192.168.141.1.4569: UDP, length 31 14:20:45.338718 IP 192.168.141.1.4569 > 192.168.141.7.4569: UDP, length 65 14:20:45.338875 IP 192.168.141.7.4569 > 192.168.141.1.4569: UDP, length 82 14:20:45.357173 IP 192.168.141.1.4569 > 192.168.141.7.4569: UDP, length 40 14:20:45.357331 IP 192.168.141.7.4569 > 192.168.141.1.4569: UDP, length 65 14:20:45.376559 IP 192.168.141.1.4569 > 192.168.141.7.4569: UDP, length 53 14:20:45.376630 IP 192.168.141.7.4569 > 192.168.141.1.4569: UDP, length 12 ^C 12 packets captured 12 packets received by filter 0 packets dropped by kernel Thelma On 06/05/2017 02:17 PM, Marcelo Terres wrote: > You can use tcpdump in your server to verify if it is receiving the > packets. > > tcpdump -ni any port 4569 > > So you have more than one ip in the server? > > On 5 Jun 2017 9:13 pm, <the...@sys-concept.com> wrote: > >> No, I don't think it is IP table issue, I've not upgraded dd-wrt for a >> while and it was zoiper was working OK with my previous version of >> asterisk. >> >> After upgrade to 11.25.1 it stop working. >> I'm sure port forwarding on dd-wrt is working OK as I have port 80 and >> 443 open. >> >> >> Thelma >> On 06/05/2017 07:12 AM, Christopher van de Sande wrote: >>> Another might be to make sure iptables isn't blocking the connection. >>> >>> You can run >>> iptables -L -n -v >>> To see if its set to block any ports. >>> >>> >>> On June 5, 2017 9:06:55 AM EDT, the...@sys-concept.com wrote: >>>> I'm getting: >>>> netstat -a |grep 4569 >>>> udp0 0 0.0.0.0:45690.0.0.0:* >>>> >>>> Should I be getting localhost IP? >>>> >>>> Thelma >>>> >>>> On 06/05/2017 06:48 AM, the...@sys-concept.com wrote: >>>>> Does asterisk listen on port 4569 by default? >>>>> >>>>> I'm running version Asterisk 11.25.1 and have a problem registering >>>>> Zoiper (IAX) to Asterisk. >>>>> I'm getting an error: >>>>> Registration refused >>>>> >>>> >>>> -- >>>> _ >>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>>> >>>> Check out the new Asterisk community forum at: >>>> https://community.asterisk.org/ >>>> >>>> New to Asterisk? Start here: >>>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >>>> >>>> asterisk-users mailing list >>>> To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: https://community.asterisk. >> org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX port 4569
No, I don't think it is IP table issue, I've not upgraded dd-wrt for a while and it was zoiper was working OK with my previous version of asterisk. After upgrade to 11.25.1 it stop working. I'm sure port forwarding on dd-wrt is working OK as I have port 80 and 443 open. Thelma On 06/05/2017 07:12 AM, Christopher van de Sande wrote: > Another might be to make sure iptables isn't blocking the connection. > > You can run > iptables -L -n -v > To see if its set to block any ports. > > > On June 5, 2017 9:06:55 AM EDT, the...@sys-concept.com wrote: >> I'm getting: >> netstat -a |grep 4569 >> udp0 0 0.0.0.0:45690.0.0.0:* >> >> Should I be getting localhost IP? >> >> Thelma >> >> On 06/05/2017 06:48 AM, the...@sys-concept.com wrote: >>> Does asterisk listen on port 4569 by default? >>> >>> I'm running version Asterisk 11.25.1 and have a problem registering >>> Zoiper (IAX) to Asterisk. >>> I'm getting an error: >>> Registration refused >>> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX port 4569
I'm getting: netstat -a |grep 4569 udp0 0 0.0.0.0:45690.0.0.0:* Should I be getting localhost IP? Thelma On 06/05/2017 06:48 AM, the...@sys-concept.com wrote: > Does asterisk listen on port 4569 by default? > > I'm running version Asterisk 11.25.1 and have a problem registering > Zoiper (IAX) to Asterisk. > I'm getting an error: > Registration refused > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX port 4569
Does asterisk listen on port 4569 by default? I'm running version Asterisk 11.25.1 and have a problem registering Zoiper (IAX) to Asterisk. I'm getting an error: Registration refused -- Thelma -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] connecting two asterisks - transfer=no
I'm using Asterisk 11 and have a problem with when making call transfer on
remote Asterisk.
This dial plan below works when I make a call directly to remote asterisk
dialing FXO on remote asterisk.
exten => 4,1,Dial(${FD_L1},25,trw)
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?vmail:line2)
exten => 4,n(line2),Dial(${FD_L2},20,rw)
exten => 4,n(vmail),Voicemail(4)
exten => 4,n,Voicemail(4)
exten => 4,n,Hangup()
Line one FD_L1 rings for 25sec. nobody pickup the line so FD_L2 rings for 20sec
and if nobody pickup the line it goes to voice mail.
However if I make a call over VPN to remote Asterisk, and dial exten: 4
FD_L1 rings for 25sec. and as soon as the call gets transferred to FD_L2 it
gets abruptly terminated with a message:
exited non-zero on 'IAX2
-- SIP/54-0006 is ringing
== Spawn extension (extensions, 4, 3) exited non-zero on 'IAX2/home_server-424'
Does it have something to do with "transfer=no" in iax.conf?
--
Thelma
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Re: [asterisk-users] Call does not go to voicemail
Tim,
I've tested similar dialplan on my home-server and it works perfectly.
(same setting, slightly different extensions) but same idea:
exten => 418,1,Dial(SIP/55,15,trw)
exten => 418,n,GotoIf($["${DIALSTATUS}"="BUSY"]?vmail:line2)
exten => 418,n(line2),Dial(SIP/218,15,rw)
exten => 418,n(vmail),Voicemail(55)
exten => 418,n,Voicemail(55)
exten => 418,n,Hangup()
I think the reason the below dialpolan IS NOT WORKING is that I'm connecting
(dialing) remote asterisk extension.
not working calling remote asterisk--
exten => 4,1,Dial(${FD_L1},25,trw)
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?vmail:line2)
exten => 4,n(line2),Dial(${FD_L2},20,rw)
exten => 4,n(vmail),Voicemail(4)
exten => 4,n,Voicemail(4)
exten => 4,n,Hangup()
-end not working calling remote asterisk-
I have two Asterisk server connected/registered over IAX and that error:
"...exited non-zero on..."
eg.
-- SIP/54-0006 is ringing
== Spawn extension (extensions, 4, 3) exited non-zero on
'IAX2/home_server-424'
I'm not the only one with this problem, this guy has the same problem as me:
http://lists.digium.com/pipermail/asterisk-users/2006-January/135612.html
--
Thelma
On 05/08/2017 06:58 PM, Tim S wrote:
> So, good, we're on the same page so far I think.
>
> As I last stated, the original code suggestion would be what you want to
> do for the serial phone ring-down (hunt), now you just need to figure
> out why your Line_2 phone is answering and then hanging up immediately
> (or why Asterisk thinks it is).
>
> I'd recommend sniffing the network traffic with Wire Shark and turning
> on some of the debug options in Asterisk to hunt down if it's the phone
> or an Asterisk quirk that is tripping up the system. We'll need more
> debug and error text to go any further with the Line_2 problem, unless
> someone much better than me can chime in with an idea... I presume
> you've already done the simple stuff like make sure your network is
> solid and that the phone firmware is up to date and stable.
>
> I'll also take a moment as an aside to suggest that you move away from
> numerical device and user names for SIP and move to text based names
> which have local meaning. The numerical names are easy to be hacked, as
> bad-guys scripts easily walk the possibilities sequentially. I find it
> also helps to use extension names in the dial plan that have meaning so
> that I can keep track of them. When a user calls an extension, the
> number they enter can feature a "Goto" with a text entry in the dial
> plan. This makes it harder for those at a phone to go places in your
> phone system they shouldn't.
>
> -Tim
>
> On Mon, May 8, 2017 at 4:51 PM, <the...@sys-concept.com
> <mailto:the...@sys-concept.com>> wrote:
>
> On 05/08/2017 04:37 PM, Tim S wrote:
> > The "error" I was talking about was in your log:
> >
> > "...== Spawn extension (extensions, 4, 3) exited non-zero on
> > 'IAX2/home_server-6364'..."
> >
> > The call terminated here in a error which prevented the dialplan from
> > continuing. Something there is broken, my recommendation is to check
> > you registrations first inside asterisk:
> >
> >> sip show peers
>
> "sip show peers" is showing FD_L2 (SIP/54 is registered)
> Name/username Host
> Dyn Forcerport ComediaACL Port Status Description
> 12(Unspecified)
> D No No 0Unmonitored
> 4/4 10.10.0.8
> D No No 5060 Unmonitored
> 54/54 10.10.0.15
> D No No 5060 Unmonitored
>
> > Something wasn't "happy" about SIP/54 in your system when Asterisk
> tried
> > talking to it.
> >
> > So you tried this:
> >
> > "...
> > Even when I put:
> > exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2)
> > exten => 4,n(line2),Dial(${FD_L2},20,trw)
> > exten => 4,n(line2),Voicemail(4)
> > ..."
> >
> > What that will do is go to the first instance of "4,n(line2)",
> which is
> > the line that seems to be triggering the channel failure. If you have
> > the Asterisk console open, I'll bet you see it spew some errors
> when you
> > try that extension routine.
> >
> > Ast
Re: [asterisk-users] Call does not go to voicemail
On 05/08/2017 04:37 PM, Tim S wrote:
> The "error" I was talking about was in your log:
>
> "...== Spawn extension (extensions, 4, 3) exited non-zero on
> 'IAX2/home_server-6364'..."
>
> The call terminated here in a error which prevented the dialplan from
> continuing. Something there is broken, my recommendation is to check
> you registrations first inside asterisk:
>
>> sip show peers
"sip show peers" is showing FD_L2 (SIP/54 is registered)
Name/username HostDyn
Forcerport ComediaACL Port Status Description
12(Unspecified)D No
No 0Unmonitored
4/4 10.10.0.8D No
No 5060 Unmonitored
54/54 10.10.0.15 D No
No 5060 Unmonitored
> Something wasn't "happy" about SIP/54 in your system when Asterisk tried
> talking to it.
>
> So you tried this:
>
> "...
> Even when I put:
> exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2)
> exten => 4,n(line2),Dial(${FD_L2},20,trw)
> exten => 4,n(line2),Voicemail(4)
> ..."
>
> What that will do is go to the first instance of "4,n(line2)", which is
> the line that seems to be triggering the channel failure. If you have
> the Asterisk console open, I'll bet you see it spew some errors when you
> try that extension routine.
>
> Asterisk dial plans are a serial processes, the first line that Asterisk
> comes across that meets the matching for a given extension and label is
> what it will run first. What you have is two lines that will match both
> extension and label - that's not really good form.
>
> My dial plan suggestion from last night would result in the functionality:
>
> Ring extension 4/Line_1, timeout 25 seconds --> if not busy then
> voicemail, else ring extension 4/Line_2, timeout 20 seconds --> voicemail.
>
>
> Again, I think you have two problems, and the bigger one is causing the
> annoying unexpected behavior in your dial plan
>
> Try doing the extension 4 without the Line_1 and see what happens:
>
> "...
> exten => 4,1,Dial(${FD_L2},20,trw)
> exten => 4,n(vmail),Voicemail(4)
> exten => 4,n,Hangup()
> ..."
I have tired the above plan with small change 4,n,Voicemail(4) (as there is no
gotoif statement)
So:
exten => 4,1,Dial(${FD_L2},20,trw)
exten => 4,n,Voicemail(4)
exten => 4,n,Hangup()
Line 2 is ring OK, and if nobody pickup the phone it goes to "Voicemail(4)" so
this part is working; there were no errors on the command line.
[snip]
But I've tired it again, this dialplan) as before and you are correct something
is wrong but command line is not showing any errors:
exten => 4,1,Dial(${FD_L1},25,trw)
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2:)
exten => 4,n(line2),Dial(${FD_L2},20,rw)
exten => 4,n,Voicemail(4)
exten => 4,n,Hangup()
I've tried:
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2)
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2:)
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?:line2)
And I get:
-- Called SIP/4
-- SIP/4-0306 is ringing
-- Nobody picked up in 25000 ms
-- Executing [4@extensions:2] GotoIf("IAX2/home_server-435", "0?line2:") in
new stack
-- Executing [4@extensions:3] Dial("IAX2/home_server-435", "SIP/54,20,rw")
in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/54
-- SIP/54-0307 is ringing
== Spawn extension (extensions, 4, 3) exited non-zero on
'IAX2/home_server-435'
-- Hungup 'IAX2/home_server-435'
So FD_L1 (exten: 4) is ringing for 25sec.; nobody pickup the phone and command
line is showing it goes to: FD_L2 (SIP/54)
-- SIP/54-0307 is ringing
but in reality FD_L2 (SIP/54) is not ringing at all, it should ring line_2 for
20sec and go to Voicemail but as soon as it prints line:
-- SIP/54-0307 is ringing
it hangs up the phone.
--
Thelma
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Re: [asterisk-users] Call does not go voicemail
Thank you for the input Tim.
Yes, that worked.
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2:vmail)
exten => 4,n(vmail),Voicemail(4)
Though, I'm not sure why are you saying line 2 is FD_L2 needs to be fixed.
Do I need to removde "t", the call can not be transferred?
Even when I put:
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2)
exten => 4,n(line2),Dial(${FD_L2},20,trw)
exten => 4,n(line2),Voicemail(4)
The call (line2) would dial "FD_L2" but would not jump to next line
"Voicemail"
--
Thelma
On 05/08/2017 12:19 AM, Tim S wrote:
> The way you have the GotoIf is making it so that no matter what the busy
> condition of the line, it will execute the next line in the dial plan.
> What you'd need is an "if" or "then" which goes to a tagged line in the
> dial plan. How it reads now is: "If [busy] then line2, else execute
> next line". Also you are saying "extension 4 is not busy", but
> extension 4 is a dialplan extension - while physical extensions "FD_L1"
> and "FD_L2" appear to be the devices which are not busy, you need to be
> clear and keep it straight in your head and text to get the best help...
>
> According to your log, nobody picked up after the 25 second timeout on
> FD_L1, so the dial status would have been NOANSWER, which would result
> in your gotoif test having a FALSE. Since you didn't specify what the
> gotoif should do if the busy test failed, it just executes the next line
> which is to call the second line (FD_L2), which it does. Then it looks
> like you have an error with the second line which causes the call to
> terminate, at which case it terminates the channel and never gets to
> voicemail.
>
>
> So it looks like two problems, 1) your FD_L2 physical extension is
> buggy, and 2) you need to label the voicemail entry point and jump to it
> if the FD_L1 was any other state but BUSY.
>
>
> "...
> exten => 4,1,Dial(${FD_L1},25,trw)
> exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2:vmail)
> exten => 4,n(line2),Dial(${FD_L2},20,trw); <--- fix me!!
> exten => 4,n(vmail),Voicemail(4)
> exten => 4,n,Hangup()
> ..."
>
>
> -Tim
>
>
> On Sun, May 7, 2017 at 9:21 PM, <the...@sys-concept.com
> <mailto:the...@sys-concept.com>> wrote:
>
> Call is not forwarded to voicemail in below dial plan, why?
>
> exten => 4,1,Dial(${FD_L1},25,trw)
> exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2)
> exten => 4,n(line2),Dial(${FD_L2},20,trw)
> exten => 4,n,Voicemail(4)
> exten => 4,n,Hangup()
>
> -- Called SIP/4
> -- SIP/4-0288 is ringing
> -- Nobody picked up in 25000 ms
> -- Executing [4@extensions:2] GotoIf("IAX2/home_server-6364",
> "0?line2") in new stack
> -- Executing [4@extensions:3] Dial("IAX2/home_server-6364",
> "SIP/54,20,trw") in new stack
> == Using SIP RTP CoS mark 5
> -- Called SIP/54
> -- SIP/54-0289 is ringing
> == Spawn extension (extensions, 4, 3) exited non-zero on
> 'IAX2/home_server-6364'
> -- Hungup 'IAX2/home_server-6364'
>
> Extension 4 is not BUSY (just nobody pickup the call) so why isn't
> call going to "Voicemail" it shouldn't ring FD_L2 (SIP/54)
> Why isn't it going to "Voicemail"?
>
> --
> Thelma
>
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[asterisk-users] Call does not go voicemail
Call is not forwarded to voicemail in below dial plan, why?
exten => 4,1,Dial(${FD_L1},25,trw)
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2)
exten => 4,n(line2),Dial(${FD_L2},20,trw)
exten => 4,n,Voicemail(4)
exten => 4,n,Hangup()
-- Called SIP/4
-- SIP/4-0288 is ringing
-- Nobody picked up in 25000 ms
-- Executing [4@extensions:2] GotoIf("IAX2/home_server-6364", "0?line2") in
new stack
-- Executing [4@extensions:3] Dial("IAX2/home_server-6364",
"SIP/54,20,trw") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/54
-- SIP/54-0289 is ringing
== Spawn extension (extensions, 4, 3) exited non-zero on
'IAX2/home_server-6364'
-- Hungup 'IAX2/home_server-6364'
Extension 4 is not BUSY (just nobody pickup the call) so why isn't call going
to "Voicemail" it shouldn't ring FD_L2 (SIP/54)
Why isn't it going to "Voicemail"?
--
Thelma
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Re: [asterisk-users] [SOLVED] configure AudioCodes MP-112 with Asterisk.
On 04/29/2017 10:16 AM, Jeff LaCoursiere wrote: > On 04/29/2017 11:12 AM, the...@sys-concept.com wrote: >> I've MP-114 that is working configured and working OK with my Asterisk >> but I just obtained MP-112 (2xFXS) and I can register OK with asterisk >> but I can only dial 3-digit extension. >> >> Anything longer than 3-digits is cut off, example I dial extension 1000: >> >> [Apr 29 10:03:30] NOTICE[3817][C-00e9]: chan_sip.c:25902 >> handle_request_invite: Call from '54' (10.0.0.115:5060) to extension >> '100' rejected because extension not found in context 'internal'. >> >> My dial plan is working OK as when I register Linsys/Sipura with >> asterisk "context=internal" and I dial any number and dial plan is >> working OK. >> >> It seems to me Audiocodes MP-112 is trimming anything that is longer >> than 3-digits. >> > Audiocodes has a default dialplan of XXX, which will trim your dialing > to three digits. Change it to X+ (or whatever makes sense). > > j Thanks for quick answer, yet that was it. The setting "Max Digits In Phone Num" was set to 3 in: Configuration->VoIP->GW and IP to IP->DTMF and Supplementary ->DTMF & Dialing I don't know when did they change it to 3 as my MP-114 came with default 32-digits Thelma -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] configure AudioCodes MP-112 with Asterisk.
I've MP-114 that is working configured and working OK with my Asterisk but I just obtained MP-112 (2xFXS) and I can register OK with asterisk but I can only dial 3-digit extension. Anything longer than 3-digits is cut off, example I dial extension 1000: [Apr 29 10:03:30] NOTICE[3817][C-00e9]: chan_sip.c:25902 handle_request_invite: Call from '54' (10.0.0.115:5060) to extension '100' rejected because extension not found in context 'internal'. My dial plan is working OK as when I register Linsys/Sipura with asterisk "context=internal" and I dial any number and dial plan is working OK. It seems to me Audiocodes MP-112 is trimming anything that is longer than 3-digits. -- Thelma -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OBi202 configure as asterisk extensions
I'm trying to configure OBi202 lines as an asterisk extension. I was able to register OBi202 to asterisk Line1 can receive a call but I can not call out via asterisk. Does anybody have a link how to configure this unit with asterisk? (Siupra/Linksys was easy to configure) this unit is different. In: ITSP Profile A DigitMap: (xx.|*xx.|#xx.) In: PHONE1 Port DigitMap: (xx.|*xx.|#xx.) When I call Line1 on OBi202 it rings but I can not make a call out neither local no outbound. -- Thelma -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
