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Can not get call forward to work.

Offline psoren

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Can not get call forward to work.
« on: July 16, 2013, 12:16:12 PM »
Hi,

I have a SME 8 server with asterisk 1.8 and SAIL 3.1.1 running in server only mode behind another SME 8 server and gateway.
They run as VM on a Vmware server

I have tried to enable CFWD in different ways, on the SIP phone (SPA 941), in the extension setup and as a call group to chose at closed hours

Nothing works, i found this line in the log file:

    NOTICE[13300] app_dial.c: Not accepting call completion offers from call-forward recipient Local/XXXXXXXX@internal-114a;1 (XXXXXXXX instead of my phone number)

I have opened ports 5060 UDPand 10000-20000 UDP in the gateway

Anyone has an idea?

Per


Offline SARK devs

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Re: Can not get call forward to work.
« Reply #1 on: July 16, 2013, 02:54:32 PM »
Hi Psoren

How did you enable the call forward?  In the SIP phone itself or by using *21*?  (I'm guessing in the phone)

Best

S

Offline bbialy

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Re: Can not get call forward to work.
« Reply #2 on: July 16, 2013, 03:05:33 PM »
Hi,
I had same problem recently and it was fixed in one of prvious versions.
current stable release is sail-3.1.1-22.noarch.rpm, which version you have?

aditionally as SARK dev wrote best practice is to use http://www.sailpbx.com/mediawiki/index.php/SAIL_Feature_Codes instead of built-in functions in IP Phones such as Call-Forward
Reading with understanding is the hardest thing IN THE WORLD

Offline psoren

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Re: Can not get call forward to work.
« Reply #3 on: July 16, 2013, 09:51:15 PM »
Hi again

*21* does not work, it activates and deactivates but result is the same. A bliip,bliip followed by occupied tone. All this was working fine in version 2, however that was not behind another SME but installed on the SME gateway itself

I noticed there was a *24* option to enable call forward, but when i try that i get the tone as number does not exist. Could the call forward function be missing?

Sail version is 3.1.1-27, asterisk is  1.8.7.0

Thanks

Per

Offline SARK devs

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Re: Can not get call forward to work.
« Reply #4 on: July 16, 2013, 09:59:11 PM »
Hi Psoren

Call forward is not missing, I promise.  :-)   Can you send me a console log of the event please?  send to admin@aelintra,com if you don't want to post it here

Best

S

Offline psoren

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Re: Can not get call forward to work.
« Reply #5 on: July 16, 2013, 10:14:56 PM »
Hi Psoren

Call forward is not missing, I promise.  :-)   Can you send me a console log of the event please?  send to admin@aelintra,com if you don't want to post it here

Hehe, good...:-)

Console log, you mean doing asterisk -r? This is what it gives:

[Jul 16 22:13:02] WARNING[24986]: file.c:653 ast_openstream_full: File pls-hold-while-try does not exist in any format
[Jul 16 22:13:02] WARNING[24986]: file.c:959 ast_streamfile: Unable to open pls-hold-while-try (format 0x8 (alaw)): No such file or directory
[Jul 16 22:13:02] WARNING[24986]: app_playback.c:475 playback_exec: ast_streamfile failed on SIP/peer1166-000000ac for pls-hold-while-try


Per

Offline bbialy

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Re: Can not get call forward to work.
« Reply #6 on: July 16, 2013, 10:39:04 PM »
Hi it is not too much
Please increase verbose level in asterisk console
core set verbose 6
And then post output.
Btw I think that you don't have installed asterisk-sounds-extra-alaw
Reading with understanding is the hardest thing IN THE WORLD

Offline psoren

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Re: Can not get call forward to work.
« Reply #7 on: July 16, 2013, 10:56:00 PM »
Hi again,

Ok i set it to 6

This i get when activating call forward:

Connected to Asterisk 1.8.7.0 currently running on sailserver (pid = 4162)
Verbosity is at least 6
  == Using SIP RTP CoS mark 5
    -- Executing [*21*XXXXXXXX@internal:1] AGI("SIP/6666-000000bd", "sarkhpe,OutCluster,*21*XXXXXXXX,,") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=Bolsje)
    -- <SIP/6666-000000bd>AGI Script sarkhpe completed, returning 0
    -- Executing [*21*XXXXXXXX@Bolsje:1] AGI("SIP/6666-000000bd", "sarkhpe,*21*XXXXXXXX,,") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Wait) Options: (0.5)
    -- AGI Script Executing Application: (Playback) Options: (call-forwarding)
[Jul 16 22:42:32] WARNING[25497]: file.c:653 ast_openstream_full: File call-forwarding does not exist in any format
[Jul 16 22:42:32] WARNING[25497]: file.c:959 ast_streamfile: Unable to open call-forwarding (format 0x1000 (g722)): No such file or directory
[Jul 16 22:42:32] WARNING[25497]: app_playback.c:475 playback_exec: ast_streamfile failed on SIP/6666-000000bd for call-forwarding
    -- AGI Script Executing Application: (Playback) Options: (activated)
    -- <SIP/6666-000000bd> Playing 'activated.gsm' (language 'da')
    -- <SIP/6666-000000bd>AGI Script sarkhpe completed, returning 0
    -- Auto fallthrough, channel 'SIP/6666-000000bd' status is 'UNKNOWN'
    -- Executing [h@Bolsje:1] Hangup("SIP/6666-000000bd", "") in new stack
  == Spawn extension (Bolsje, h, 1) exited non-zero on 'SIP/6666-000000bd'

This is when i call to the number that should be forwarded:

Connected to Asterisk 1.8.7.0 currently running on sailserver (pid = 4162)
Verbosity is at least 6
  == Using SIP RTP CoS mark 5
    -- Executing [75101234@mainmenu:1] AGI("SIP/peer1166-000000c1", "sarkhpe,Inbound,75101234,,") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=75101234)
    -- <SIP/peer1166-000000c1>AGI Script sarkhpe completed, returning 0
    -- Executing [6666@extensions:1] AGI("SIP/peer1166-000000c1", "sarkhpe,InCall,,,") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Playback) Options: (silence/1)
    -- <SIP/peer1166-000000c1> Playing 'silence/1.alaw' (language 'da')
    -- AGI Script Executing Application: (Playback) Options: (pls-hold-while-try)
[Jul 16 22:47:53] WARNING[25580]: file.c:653 ast_openstream_full: File pls-hold-while-try does not exist in any format
[Jul 16 22:47:53] WARNING[25580]: file.c:959 ast_streamfile: Unable to open pls-hold-while-try (format 0x8 (alaw)): No such file or directory
[Jul 16 22:47:53] WARNING[25580]: app_playback.c:475 playback_exec: ast_streamfile failed on SIP/peer1166-000000c1 for pls-hold-while-try
    -- AGI Script Executing Application: (Set) Options: (CALLERID(rdnis)=6666)
    -- <SIP/peer1166-000000c1>AGI Script sarkhpe completed, returning 0
    -- Executing [XXXXXXXX@internal:1] AGI("SIP/peer1166-000000c1", "sarkhpe,OutCluster,XXXXXXXX,,") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=6666)
    -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=Bolsje)
    -- <SIP/peer1166-000000c1>AGI Script sarkhpe completed, returning 0
    -- Executing [XXXXXXXX@Bolsje:1] AGI("SIP/peer1166-000000c1", "sarkhpe,OutRoute,BK,,") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Set) Options: (TIMEOUT(absolute)=3600)
Channel will hangup at 2013-07-16 23:47:53.329 CEST.
    -- AGI Script Executing Application: (Dial) Options: (SIP/XXXXXXXX@peer2065,,Tr)
  == Using SIP RTP CoS mark 5
    -- Called SIP/XXXXXXXX@peer2065
    -- Got SIP response 503 "Service Unavailable" back from 178.20.218.200:5060
    -- SIP/peer2065-000000c2 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- AGI Script Executing Application: (Playback) Options: (beep)
    -- <SIP/peer1166-000000c1> Playing 'beep.alaw' (language 'da')
    -- AGI Script Executing Application: (Set) Options: (TIMEOUT(absolute)=3600)
Channel will hangup at 2013-07-16 23:47:53.847 CEST.
    -- AGI Script Executing Application: (Dial) Options: (SIP/XXXXXXXX@peer8353,,Tr)
  == Using SIP RTP CoS mark 5
    -- Called SIP/XXXXXXXX@peer8353
    -- Got SIP response 503 "Service Unavailable" back from 178.20.218.200:5060
    -- SIP/peer8353-000000c3 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- AGI Script Executing Application: (Playback) Options: (beep)
    -- <SIP/peer1166-000000c1> Playing 'beep.alaw' (language 'da')
    -- AGI Script Executing Application: (Playtones) Options: (congestion)
    -- AGI Script Executing Application: (Congestion) Options: ()
    -- <SIP/peer1166-000000c1>AGI Script sarkhpe completed, returning 4
  == Spawn extension (Bolsje, XXXXXXXX, 1) exited non-zero on 'SIP/peer1166-000000c1'
    -- Executing [h@Bolsje:1] Hangup("SIP/peer1166-000000c1", "") in new stack
  == Spawn extension (Bolsje, h, 1) exited non-zero on 'SIP/peer1166-000000c1'

Actually, doesn't this look like the SIP provider is causing the problem?

Got SIP response 503 "Service Unavailable" back from 178.20.218.200":5060


Per

Offline bbialy

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Re: Can not get call forward to work.
« Reply #8 on: July 16, 2013, 11:52:25 PM »
I think the clue is
    -- Got SIP response 503 "Service Unavailable" back from 178.20.218.200:5060
Probably the problem is related with CALLERID.

Situation num A calls to num B. Num B sets call forward to num C.

So your sip provider receive call: Num A calls num C.
But Num A is not valid Number for your service so it may be blocked by anti fraud mechanisms

To check if am I right please do test only on internal extensions.

If I am right you can try to set outbound CALLERID in trunk web panel.
Additionally you can try to add fromuser=[your_Sip_number]
Reading with understanding is the hardest thing IN THE WORLD

Offline psoren

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Re: Can not get call forward to work.
« Reply #9 on: July 17, 2013, 10:47:02 AM »
I think the clue is
    -- Got SIP response 503 "Service Unavailable" back from 178.20.218.200:5060
Probably the problem is related with CALLERID.

To check if am I right please do test only on internal extensions.

If I am right you can try to set outbound CALLERID in trunk web panel.
Additionally you can try to add fromuser=[your_Sip_number]

Ok i tried these things but they make no difference.

Actually this worked just fine in Sail-2.6.1-11 and asterisk 1.4.42 on my SME 7.5 just 3 weeks ago on the old server

I will ask the SIP provider if they made changes.

Per

Offline psoren

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Re: Can not get call forward to work.
« Reply #10 on: July 17, 2013, 05:19:46 PM »
Apparantly there are som issues with the SIP provider, i am trying to work it out with them and then post the result and reason here.

Thanks
Per

Offline bbialy

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Re: Can not get call forward to work.
« Reply #11 on: July 17, 2013, 11:50:32 PM »
Hi,
what are the results when you test on internal extensions only?

if on internal extensions is OK it means that problem is with your sip provider other way problem is with sail.

IMHO peanuts against dollars that on internal extensions will work fine.

did you tried fromuser setting in your trunk definition?
fromuser=[your_ddi_number]

additionally please check what are you sending in from field in sip protocol during divert.
command on asterisk console:
sip set debug peer [peername]

it will HUGE amount of text but you are looking something like INVITE and then try to find From:
From: "5444545645" <sip:5444545645@91..XXX.XXX.XX>;tag=as0b1a5163
Code: [Select]
INVITE sip:48XXXXXXX@sip.XXX.pl SIP/2.0
Via: SIP/2.0/UDP 91.XXX.XXX.XX:5060;branch=z9hG4bK46256266
Max-Forwards: 70
From: "5444545645" <sip:5444545645@91..XXX.XXX.XX>;tag=as0b1a5163
To: <sip:48XXXXXXX@sip.XXX.pl>
Contact: <sip:5444545645@91..XXX.XXX.XX:5060>
Call-ID: 6272275438b7b6e731cbefc7550b3679@91..XXX.XXX.XX:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.7.0
Date: Wed, 17 Jul 2013 21:44:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 367177668 367177668 IN IP4 91.239.254.37
s=Asterisk PBX 1.8.7.0
c=IN IP4 91..XXX.XXX.XX
t=0 0
m=audio 10922 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Reading with understanding is the hardest thing IN THE WORLD

Offline psoren

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Re: Can not get call forward to work.
« Reply #12 on: July 18, 2013, 10:59:16 PM »
Apparantly there are som issues with the SIP provider, i am trying to work it out with them and then post the result and reason here.

Thanks
Per

OK, it is working now.

My SIP provider changed some settings on their server which allowed it to call forward with extension caller id CALLERID(rdnis)

Apparantly version 3 send local extension but version 2 not. Which is good and which is bad i have no clue ;-)

Thanks for the help.

Per