[asterisk-users] problem transferring calls some of the times
Ian
asterisk at iancoetzee.za.net
Fri Feb 22 03:47:31 CST 2008
Hi All
Agter a bit of logging to a syslog server, I found a peculiar entry
today, ironically right after a call failed to transfer. They key
sequence and call path used until it gets transferred is as follows
* Phone rings on Asterisk
* Asterisk transferres to the receptionists phone (GXP 2000)
o Receptionist doensnt answer for 15 seconds, and the call
gets routed to the bosses secrataries phone
* Bosses secretary answers the phone and tries to transfer it to the
boss with the keysequence "flash", extention "315", talks,
"transfer" but the transfer is the one that fails
Message in the log is
> Feb 22 09:55:22 10.219.127.102 GS_LOG:
> [00:0B:82:13:02:CF][000][FFFD][01010412] Received SIP message: 407
> Feb 22 09:55:22 10.219.127.102 GS_LOG:
> [00:0B:82:13:02:CF][000][FFFD][01010412] SIP dialog matched to channel 0
> Feb 22 09:55:22 10.219.127.102 GS_LOG:
> [00:0B:82:13:02:CF][000][FFFD][01010412] Send SIP message: ACK To
> 10.219.127.7:5060, sip_handle: 0x0046F09C
> Feb 22 09:55:22 10.219.127.102 GS_LOG:
> [00:0B:82:13:02:CF][000][FFFD][01010412] sip_len: 553, sip_handle:
> 0x0046F09C, ACK sip:00123463409 at 10.219.127.7;user=p
> hone SIP/2.0 Via: SIP/2.0/UDP
> 10.219.127.102:5060;branch=z9hG4bK623e473ec5e8c5e8 From: "Wanda"
> <sip:312 at 10.219.127.7;user=phone>;tag=1a7b934ecd3e23f7 To:
> <sip:00123463409 at 10.219.127.7;user=phone>;tag=as07aa3c42 Contact:
> <sip:312 at 10.219.127.102:5060;transport=udp;user=phone> Supported:
> path Call-ID: 1138f5f7
> 40c8d06c at 10.219.127.102 CSeq: 58150 ACK User-Agent: Grandstream
> BT200 1.1.4.18 Max-Forwards: 70 Allow:
> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU
> BSCRIBE,UPDATE,PRACK Content-Length: 0
The message that got my worried is the one saying "Recieved SIP message
407", can that be the ghost I am looking for?
Extentions.conf
[incoming_calls]
exten => s,1,NoOp(${CALLERID(name)} skakel Luzaan)
exten => s,n,dial(SIP/300,15)
exten => s,n,Set(CALLERID(name)=deur)
exten => s,n,Set(CALLERID(num)=deur)
exten => s,n,NoOp(${CALLERID(name)} skakel Wanda)
exten => s,n,dial(SIP/312)
;exten => s,n,dial(SIP/317)
exten => s,n,Hangup()
[internal]
exten => 900,1,Verbose(1|Echo test application)
exten => 900,n,Echo()
exten => 900,n,Hangup()
;interne oproepe
exten => _3XX,1,NoOp(${CALLERID} skakel ${EXTEN})
exten => _3XX,n,Dial(SIP/${EXTEN},30)
;exten => _3XX,n,execif(${CALLERID} != _3XX|goto|incoming_calls/s/1)
exten => _3XX,n,goto(incoming_calls,s,1)
exten => _3XX,n,Hangup()
Sip.conf
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
limitonpeers=yes
allowtransfer=yes
callevents=yes
regcontext=GXP_BLF
[sets](!)
type=friend
context=internal
host=dynamic
;disallow=all
;allow=speex
secret=test
dtmfmode=info
callgroup=1
pickupgroup=1
call-limit=20
subscribecontext=GXP_BLF
canreinvite=yes
nat=no
[300](sets) ;Luzaan
regexten=300
Any help will be apreciated
Thanks
Ian
Ian said the following on 22-Feb-08 10:06 AM:
> Hi,
>
> Mojo with Horan & Company, LLC said the following on 20-Feb-08 09:31 PM:
>> Is it AFTER you have parked a call? Meaning, for example, you transfer
>> an incoming call to 700. No problem. Later, when it's picked up from
>> 701, can it NOT be transferred again?
>>
>> Moj
>>
> No I don't park the call.
>
> The call comes in, and gets redirected to our receptionists phone,
> from there it gets transferred to another extension (the bosses
> secratary) and then gets transferred (to the boss). now the problem,
> sometimes that transfer fails, other times the call dont even want to
> leave the receptionists phone.
>
> The big thing about this problem is that it comes and goes, like
> yesterday we didn't have a problem, and I did not change a thing.
>
> Ian
>> Ian wrote:
>>
>>> Hi All
>>>
>>> Sorry to be a bother again but seems like I just cant get away from
>>> the problems.
>>>
>>> This time my problem is that *sometimes* a user cant transfer a call
>>> from one extension to another, I have narrowed down the problem to it
>>> only happening to calls from outside the internal system.
>>>
>>> The wierd thing about the problem is that it comes and goes one moment
>>> the user can transfer, and the next call he can't.
>>>
>>> I am running:
>>>
>>> * Asterisk 1.4.17
>>> * Zaptel 1.4.7.1
>>> * Libpri 1.4.3
>>>
>>> Using the following phones and firmware
>>>
>>> * Grandstream GXP2000 (with ext pad) : 1.1.4.14
>>> * Grandstream BT200 : 1.1.4.18
>>>
>>> I have set up the phones to log debug logs to a syslog server, I am
>>> still trying to figure out what exactly the log says.
>>>
>>> Is it an * problem, or Grandstream problem
>>>
>>> Does anyone know if I am able to see the keysequence the user types
>>> into the phone (just in case it might even be a user made problem), I
>>> have tried scanning though the logs of a failed call, but could not
>>> see any lines that can be a keypress, or maybe I am looking in the
>>> incorrect spot?
>>>
>>> Your help will be greatly appreciated.
>>>
>>> Let me know if, in any way, I can shed some more light on the subject.
>>>
>>> Thanks in advance
>>> Ian
>>> --
>>> www.vddi.co.za <http://www.vddi.co.za/>
>>> I Coetzee
>>> IT Tegnikus
>>> Telefoon : 012 664 2300
>>> Selfoon : 079 522 6519
>>> Faks : 012 644 2902
>>> E-pos : ian at vddi.co.za
>>> Skype : vddb_igcoetzee
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> 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 --
>>
>> 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 --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080222/6a7bc62f/attachment.htm
More information about the asterisk-users
mailing list