No subject

Tue Sep 5 14:32:44 MST 2006

Regards,

Daniel

WipeOut . a écrit:

>Just add callgroup={number} and pickupgroup={number} into each SIP phone's config in the sip.conf file..
>
>  
>
>>Hello,
>>
>>What configuration should I use for this (I use sip phones)?
>>
>>Best regards,
>>
>>Daniel
>>
>>
>>WipeOut . a ?crit:
>>
>>    
>>
>>>OK you are correct..
>>>
>>>*8 picks up the call..I wonder why *8# does not work??
>>>
>>>I also had the same problem that the phone that I collected the call from did not stop ringing..
>>>
>>>
>>> 
>>>
>>>      
>>>
>>>>I have problems with this as well ( similar config ).  My CVS is 10 days 
>>>>old.
>>>>
>>>>I can get the call picked up with *8     (  *8# does not work )  but 
>>>>the phone B never stops ringing.
>>>>B rings forever. I'm using SNOM200.
>>>>
>>>>--Pertti
>>>>
>>>>
>>>>WipeOut . wrote:
>>>>
>>>>   
>>>>
>>>>        
>>>>
>>>>>I have just started to play with callgroups and pickupgroups..
>>>>>
>>>>>I updates my * from CVS this morning (about 15 mins ago)..
>>>>>
>>>>>I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf..
>>>>>
>>>>>I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB..
>>>>>
>>>>>Have I not configured somthing correctly or is there a bug??
>>>>>
>>>>>Later.
>>>>>
>>>>>
>>>>>     
>>>>>
>>>>>          
>>>>>
>>>>-- 
>>>>
>>>>**********************************************************************
>>>>Nordic LAN&WAN Communication Oy
>>>>Pertti Pikkarainen
>>>>vp of engineering
>>>>E-Mail: ppik at lanwan.fi
>>>>tel: +358-9-5024100
>>>>fax: +358-9-5023840
>>>>gsm: +358-500-511467
>>>>
>>>>Sinikalliontie 16
>>>>02630 Espoo
>>>>FINLAND
>>>>
>>>>**********************************************************************
>>>>
>>>>
>>>>
>>>>_______________________________________________
>>>>Asterisk-Users mailing list
>>>>Asterisk-Users at lists.digium.com
>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>   
>>>>
>>>>        
>>>>
>>> 
>>>
>>>      
>>>
>>-- 
>>Daniel ANDRE (mailto:dandre at iris-tech.fr)
>>IRIS Technologies - http://www.iris-tech.com
>>Serveur kwartz - http://www.kwartz.com
>>
>>    
>>
>
>  
>

-- 
Daniel ANDRE (mailto:dandre at iris-tech.fr)
IRIS Technologies - http://www.iris-tech.com
Serveur kwartz - http://www.kwartz.com

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It's exactly what I have done, I have this log message: 
NOTICE[114696]: File chan_sip.c, Line 4870 (handle_request): Nothing to
pick up 
WARNING[114696]: File chan_sip.c, Line 2220 (__transmit_response):
Unable to determine sequence number from '' 
 
From&nbsp; 276 I dial 326 and trie to pickup using&nbsp; 273. I include my
sip.conf 
 
Regards, 
 
Daniel 
 
 
WipeOut . a &eacute;crit: 
<blockquote type="cite"
 cite="mid20030909101244.26701.qmail at linuxmail.org">
 <pre wrap="">Just add callgroup={number} and pickupgroup={number} into each SIP phone's config in the sip.conf file..

  </pre>
  <blockquote type="cite">
    <pre wrap="">Hello,

What configuration should I use for this (I use sip phones)?

Best regards,

Daniel

WipeOut . a ?crit:

    </pre>
    <blockquote type="cite">
      <pre wrap="">OK you are correct..

*8 picks up the call..I wonder why *8# does not work??

I also had the same problem that the phone that I collected the call from did not stop ringing..

      </pre>
      <blockquote type="cite">
        <pre wrap="">I have problems with this as well ( similar config ).  My CVS is 10 days 
old.

I can get the call picked up with *8     (  *8# does not work )  but 
the phone B never stops ringing.
B rings forever. I'm using SNOM200.

--Pertti

WipeOut . wrote:

        </pre>
        <blockquote type="cite">
          <pre wrap="">I have just started to play with callgroups and pickupgroups..

I updates my * from CVS this morning (about 15 mins ago)..

I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf..

I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB..

Have I not configured somthing correctly or is there a bug??

Later.

          </pre>
        </blockquote>
        <pre wrap="">-- 

**********************************************************************
Nordic LAN&amp;WAN Communication Oy
Pertti Pikkarainen
vp of engineering
E-Mail: <a class="moz-txt-link-abbreviated" href="mailto:ppik at lanwan.fi">ppik at lanwan.fi</a>
tel: +358-9-5024100
fax: +358-9-5023840
gsm: +358-500-511467

Sinikalliontie 16
02630 Espoo
FINLAND

**********************************************************************

_______________________________________________
Asterisk-Users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Asterisk-Users at lists.digium.com">Asterisk-Users at lists.digium.com</a>
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>

        </pre>
      </blockquote>
      <pre wrap=""> 

      </pre>
    </blockquote>
    <pre wrap="">-- 
Daniel ANDRE (<a class="moz-txt-link-freetext" href="mailto:dandre at iris-tech.fr">mailto:dandre at iris-tech.fr</a>)
IRIS Technologies - <a class="moz-txt-link-freetext" href="http://www.iris-tech.com">http://www.iris-tech.com</a>
Serveur kwartz - <a class="moz-txt-link-freetext" href="http://www.kwartz.com">http://www.kwartz.com</a>

    </pre>
  </blockquote>
  <pre wrap=""><!---->
  </pre>
</blockquote>
<br>
<pre class="moz-signature" cols="72">-- 
Daniel ANDRE (<a class="moz-txt-link-freetext" href="mailto:dandre at iris-tech.fr">mailto:dandre at iris-tech.fr</a>)
IRIS Technologies - <a class="moz-txt-link-freetext" href="http://www.iris-tech.com">http://www.iris-tech.com</a>
Serveur kwartz - <a class="moz-txt-link-freetext" href="http://www.kwartz.com">http://www.kwartz.com</a>
</pre>
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 name="sip.conf"
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 filename="sip.conf"

;
; SIP Configuration for Asterisk
;
[general]
port = 5060			; Port to bind to
bindaddr = 192.168.10.254	; Address to bind to
context = SIP			; Default for incoming calls
tos=lowdelay
;tos=184
;tos=50
;rxgain=30
;txgain=30
threewaycalling=yes

allow=ALAW
disallow=GSM
disallow=ULAW

; valeurs pas defaut

;téléphone grandstream benoit.
[276]
mailbox=276
type = friend
host = dynamic
canreinvite = yes
dtmf=inband
pickupgroup=1
callgroup=1

[273] ;téléphone grandstream Antoine
mailbox=273
type = friend
host = dynamic
canreinvite = yes
dtmf=inband
pickupgroup=1
callgroup=1

[235] ; téléphone grandstream Arnaud
mailbox=235
type = friend
host = dynamic
canreinvite = yes
dtmf=inband
pickupgroup=1
callgroup=1

[338] ;téléphone grandstream Dominique
mailbox=338
type = friend
host = dynamic
canreinvite = yes
dtmf=inband
pickupgroup=1
callgroup=1

[222] ; téléphone grandstream Daniel
mailbox=326
type = friend
host = 192.168.0.2
canreinvite = yes
dtmf=inband

[326] ; téléphone grandstream Daniel
mailbox=326
type = friend
host = dynamic
canreinvite = yes
dtmf=inband
pickupgroup=1
callgroup=1

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