RE: [Asterisk-Users] Grandstream CallerID
I've been working with Grandstream support on the caller ID thing and after
looking at a packet capture using Ethereal, it seems that * is sending the
characters to the phone rather than the number. I've got the following
configured in the extensions.conf file:
exten => s,1,SetCallerID(${CALLERIDNUM}); Set the caller ID
exten => s,2,Dial(SIP/1234,15,tTrf) ; Dial our office SIP phone
When I look at the logs, it says that its setting the CID to the number but
Ethereal says otherwise. Any ideas?
Thanks,
Dave
-Original Message-
From: Wilson Pickett [mailto:[EMAIL PROTECTED]
Sent: Sunday, December 19, 2004 2:29 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Grandstream CallerID
> I forgot to ask, since the BT100 can't take characters (only numbers), I
> would have assumed that there was a function to extract a number from an
> incoming PSTN CID, is that possible?
Try this
exten => s,5,SetCIDNum(1234)
and see if the phone displays it
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Re: [Asterisk-Users] Grandstream CallerID
David Ishmael wrote:
I reloaded using asterisk -rx reload.
-Dave
Finally, I could make it work with setup below. I recommend you try to
restart (not just reload) asterisk and your phone after you changed the
configuration.
extension.conf snippet:
[sip_phones]
exten => _ZX,1,NoOp(CALLERID=${CALLERID})
exten => _ZX,2,SetCIDNum(${CALLERIDNUM})
exten => _ZX,3,Dial(SIP/ZX)
and sip.conf snippet:
[42]
type=friend
host=dynamic; we have a static but private IP address
username=42
secret=
callerid=Philipp Ebneter <42>
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Re: [Asterisk-Users] Grandstream CallerID
I use SetCallerID() and it displays the number just fine on my
GrandStream phone.
Regards,
Scott Stingel
www.evtmedia.com
David Ishmael wrote:
I'm confused, should I be using SetCallerID(${CALLERIDNUM}) or
SetCIDNum(${CALLERIDNUM})? Also, I don't think it matters but I'm trying to
forward the CID coming in from the PSTN line. I know Asterisk sees the CID
because its shows up in the logs. I think I've tried just about every
combination possible and have pretty much given up hope on getting the GS
phone to display the CID.
-Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
Ebneter
Sent: Tuesday, December 21, 2004 6:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Grandstream CallerID
Those suggestions were just for testing, obviously.
iax.conf
[2000]
callerid="My Name" <2000>
sip.conf
[2002]
type=friend
username=2002
Dialing 2002 from 2000
-- Executing Dial("IAX2/[EMAIL PROTECTED]/3", "SIP/2002|20|tT") in new stack
2000 appears on the BT100
___
hmmm... interesting. I made no changes to the config, but after this
morning when we had to restart the server my phone now actually
displayes a correct number set with the SETCIDNUM.
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RE: [Asterisk-Users] Grandstream CallerID
I reloaded using asterisk -rx reload.
-Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer
Sent: Tuesday, December 21, 2004 7:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Grandstream CallerID
Sorry about the question, but did you reload asterisk after making the
changes to your conf files?
> -Original Message-
> From: Philipp Ebneter [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, December 21, 2004 2:04 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Grandstream CallerID
>
> Those suggestions were just for testing, obviously.
>
> >iax.conf
> >[2000]
> >callerid="My Name" <2000>
> >
> >sip.conf
> >[2002]
> >type=friend
> >username=2002
> >
> >Dialing 2002 from 2000
> >
> >-- Executing Dial("IAX2/[EMAIL PROTECTED]/3", "SIP/2002|20|tT") in new stack
> >
> >2000 appears on the BT100
> >___
> >
> hmmm... interesting. I made no changes to the config, but after this
> morning when we had to restart the server my phone now actually
> displayes a correct number set with the SETCIDNUM.
>
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RE: [Asterisk-Users] Grandstream CallerID
I'm confused, should I be using SetCallerID(${CALLERIDNUM}) or
SetCIDNum(${CALLERIDNUM})? Also, I don't think it matters but I'm trying to
forward the CID coming in from the PSTN line. I know Asterisk sees the CID
because its shows up in the logs. I think I've tried just about every
combination possible and have pretty much given up hope on getting the GS
phone to display the CID.
-Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
Ebneter
Sent: Tuesday, December 21, 2004 6:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Grandstream CallerID
Those suggestions were just for testing, obviously.
>iax.conf
>[2000]
>callerid="My Name" <2000>
>
>sip.conf
>[2002]
>type=friend
>username=2002
>
>Dialing 2002 from 2000
>
>-- Executing Dial("IAX2/[EMAIL PROTECTED]/3", "SIP/2002|20|tT") in new stack
>
>2000 appears on the BT100
>___
>
hmmm... interesting. I made no changes to the config, but after this
morning when we had to restart the server my phone now actually
displayes a correct number set with the SETCIDNUM.
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RE: [Asterisk-Users] Grandstream CallerID
Sorry about the question, but did you reload asterisk after making the
changes to your conf files?
> -Original Message-
> From: Philipp Ebneter [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, December 21, 2004 2:04 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Grandstream CallerID
>
> Those suggestions were just for testing, obviously.
>
> >iax.conf
> >[2000]
> >callerid="My Name" <2000>
> >
> >sip.conf
> >[2002]
> >type=friend
> >username=2002
> >
> >Dialing 2002 from 2000
> >
> >-- Executing Dial("IAX2/[EMAIL PROTECTED]/3", "SIP/2002|20|tT") in new stack
> >
> >2000 appears on the BT100
> >___
> >
> hmmm... interesting. I made no changes to the config, but after this
> morning when we had to restart the server my phone now actually
> displayes a correct number set with the SETCIDNUM.
>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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> dangerous content by MailScanner, and is
> believed to be clean.
> MailScanner thanks transtec Computers for their support.
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Re: [Asterisk-Users] Grandstream CallerID
Those suggestions were just for testing, obviously.
iax.conf
[2000]
callerid="My Name" <2000>
sip.conf
[2002]
type=friend
username=2002
Dialing 2002 from 2000
-- Executing Dial("IAX2/[EMAIL PROTECTED]/3", "SIP/2002|20|tT") in new stack
2000 appears on the BT100
___
hmmm... interesting. I made no changes to the config, but after this
morning when we had to restart the server my phone now actually
displayes a correct number set with the SETCIDNUM.
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Re: [Asterisk-Users] Grandstream CallerID
> I have the same problem as David. No matter what I tried, I always see
> the username of extension (the phone) himself.
> In the following example, when I get a call from somebody, I just see
> "42" (=my number), but not who's calling me.
>
> Setting the CallerID /CallerIdNum doen't work either:
> -- Executing SetCIDNum("CAPI[contr2/0142]/113", "1234") in new stack
Those suggestions were just for testing, obviously.
iax.conf
[2000]
callerid="My Name" <2000>
sip.conf
[2002]
type=friend
username=2002
Dialing 2002 from 2000
-- Executing Dial("IAX2/[EMAIL PROTECTED]/3", "SIP/2002|20|tT") in new stack
2000 appears on the BT100
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Re: [Asterisk-Users] Grandstream CallerID
Try this
exten => s,5,SetCIDNum(1234)
and see if the phone displays it
I have the same problem as David. No matter what I tried, I always see
the username of extension (the phone) himself.
In the following example, when I get a call from somebody, I just see
"42" (=my number), but not who's calling me.
Setting the CallerID /CallerIdNum doen't work either:
-- Executing SetCIDNum("CAPI[contr2/0142]/113", "1234") in new stack
==> has no effect
Sip.conf snippet:
[general]
context=from_sip
port=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no
incominglimit=2
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g723.1
allow=g729
[42]
type=friend
host=dynamic
username=42
secret=
dtmfmode=rfc2833
callerid=Philipp Ebneter <42>
callgroup=1
pickupgroup=1
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Re: [Asterisk-Users] Grandstream CallerID
> I forgot to ask, since the BT100 can't take characters (only numbers), I > would have assumed that there was a function to extract a number from an > incoming PSTN CID, is that possible? Try this exten => s,5,SetCIDNum(1234) and see if the phone displays it ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream CallerID
I forgot to ask, since the BT100 can't take characters (only numbers), I
would have assumed that there was a function to extract a number from an
incoming PSTN CID, is that possible?
Thanks again,
David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Ishmael
Sent: Sunday, December 19, 2004 2:03 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Grandstream CallerID
I did have fromuser set in my sip.conf so I went in and commented the line
out (thanks for the help on that). This is what I have in my
extensions.conf file:
exten => s,1,SetCallerID(${CALLERID}) ; Set the caller ID
exten => s,2,Wait(2)
exten => s,3,Dial(SIP/1234,20,tr) ; Dial our office SIP phone
exten => s,4,Hangup
That produces the full CID coming from the PSTN (Verizon) in the log
something like:
"Joe Somebody" <7035551212>
When I change the SetCallerID() function to:
exten => s,1,SetCallerID(111) ; Set the caller ID
It still doesn't send the CID to the phone (or if it does, the phone isn't
recognizing it). Do I have my extensions.conf file setup wrong? Is there
something special I need to do in the SIP phone configuration? Here's the
SIP section of the /etc/asterisk/sip.conf:
[1234]
type=friend
context=sip
username=1234
secret=
callerid=Office <1234>
host=dynamic
nat=no
canreinvite=yes
dtmfmode=inband
incominglimit=1
[EMAIL PROTECTED]
disallow=all
allow=ulaw
allow=alaw
I noticed on the WiKi there's a function called SetCIDNum(), but that
doesn't do anything (or at least nothing I could see). I'm stumped on this
one. Any advice is always welcomed. :)
Thanks,
David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Clauson
Sent: Sunday, December 19, 2004 11:44 AM
To: Wilson Pickett; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Grandstream CallerID
On Sunday 19 December 2004 06:31, Wilson Pickett wrote:
> > Is it possible to send the incoming PSTN caller ID to a Grandstream
> > Budge Tone-100 SIP phone? I've configured the extensions.conf file
> > and the log is
>
> As Eric notes, the BT100 phones won't show letters. If a call comes
> in without CID, asterisk sends a string like "Asterisk call" which
> the BT will try to display as some giberish so I have setcallerid to
> "000" when this happens. I'd recommend using setcallerid(1234567890)
> (or any number) to test the phone, which should display that. If
> callerid does come in from PSTN, it should just make through as he
> said.
I'm having the same issue as David, and forcing the CallerID with
SetCallerID() doesn't work - it still only shows the extension of the
phone I'm calling. (Yes, I've checked the CDR and also done a show
channel on the phone while ringing - the Caller ID is reset properly
I think I have the solution -- I had the fromuser= variable set, which
is what Asterisk uses to force CallerID as when it's making calls out
to the Grandstream. Is that set, David? If so, unset it, and you
should be fine.
--mec
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RE: [Asterisk-Users] Grandstream CallerID
I did have fromuser set in my sip.conf so I went in and commented the line
out (thanks for the help on that). This is what I have in my
extensions.conf file:
exten => s,1,SetCallerID(${CALLERID}) ; Set the caller ID
exten => s,2,Wait(2)
exten => s,3,Dial(SIP/1234,20,tr) ; Dial our office SIP phone
exten => s,4,Hangup
That produces the full CID coming from the PSTN (Verizon) in the log
something like:
"Joe Somebody" <7035551212>
When I change the SetCallerID() function to:
exten => s,1,SetCallerID(111) ; Set the caller ID
It still doesn't send the CID to the phone (or if it does, the phone isn't
recognizing it). Do I have my extensions.conf file setup wrong? Is there
something special I need to do in the SIP phone configuration? Here's the
SIP section of the /etc/asterisk/sip.conf:
[1234]
type=friend
context=sip
username=1234
secret=
callerid=Office <1234>
host=dynamic
nat=no
canreinvite=yes
dtmfmode=inband
incominglimit=1
[EMAIL PROTECTED]
disallow=all
allow=ulaw
allow=alaw
I noticed on the WiKi there's a function called SetCIDNum(), but that
doesn't do anything (or at least nothing I could see). I'm stumped on this
one. Any advice is always welcomed. :)
Thanks,
David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Clauson
Sent: Sunday, December 19, 2004 11:44 AM
To: Wilson Pickett; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Grandstream CallerID
On Sunday 19 December 2004 06:31, Wilson Pickett wrote:
> > Is it possible to send the incoming PSTN caller ID to a Grandstream
> > Budge Tone-100 SIP phone? I've configured the extensions.conf file
> > and the log is
>
> As Eric notes, the BT100 phones won't show letters. If a call comes
> in without CID, asterisk sends a string like "Asterisk call" which
> the BT will try to display as some giberish so I have setcallerid to
> "000" when this happens. I'd recommend using setcallerid(1234567890)
> (or any number) to test the phone, which should display that. If
> callerid does come in from PSTN, it should just make through as he
> said.
I'm having the same issue as David, and forcing the CallerID with
SetCallerID() doesn't work - it still only shows the extension of the
phone I'm calling. (Yes, I've checked the CDR and also done a show
channel on the phone while ringing - the Caller ID is reset properly
I think I have the solution -- I had the fromuser= variable set, which
is what Asterisk uses to force CallerID as when it's making calls out
to the Grandstream. Is that set, David? If so, unset it, and you
should be fine.
--mec
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Re: [Asterisk-Users] Grandstream CallerID
On Sunday 19 December 2004 06:31, Wilson Pickett wrote: > > Is it possible to send the incoming PSTN caller ID to a Grandstream > > Budge Tone-100 SIP phone? I've configured the extensions.conf file > > and the log is > > As Eric notes, the BT100 phones won't show letters. If a call comes > in without CID, asterisk sends a string like "Asterisk call" which > the BT will try to display as some giberish so I have setcallerid to > "000" when this happens. I'd recommend using setcallerid(1234567890) > (or any number) to test the phone, which should display that. If > callerid does come in from PSTN, it should just make through as he > said. I'm having the same issue as David, and forcing the CallerID with SetCallerID() doesn't work - it still only shows the extension of the phone I'm calling. (Yes, I've checked the CDR and also done a show channel on the phone while ringing - the Caller ID is reset properly I think I have the solution -- I had the fromuser= variable set, which is what Asterisk uses to force CallerID as when it's making calls out to the Grandstream. Is that set, David? If so, unset it, and you should be fine. --mec ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream CallerID
> Is it possible to send the incoming PSTN caller ID to a Grandstream Budge > Tone-100 SIP phone? I've configured the extensions.conf file and the log is As Eric notes, the BT100 phones won't show letters. If a call comes in without CID, asterisk sends a string like "Asterisk call" which the BT will try to display as some giberish so I have setcallerid to "000" when this happens. I'd recommend using setcallerid(1234567890) (or any number) to test the phone, which should display that. If callerid does come in from PSTN, it should just make through as he said. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream CallerID
David Ishmael wrote: Is it possible to send the incoming PSTN caller ID to a Grandstream Budge Tone-100 SIP phone? I've configured the extensions.conf file and the log is definitely showing the incoming caller ID information, but its not being sent to the SIP phone (assuming that's possible). I've searched through the mailing list threads looking for an answer to this but came up empty handed. The BT101 only supports NUMBER, not NAME. Usually CallerID just works if Asterisk sees it. -- I am seeking part or full time employment in the Greater Toronto Area, My preference is part time employment with some telecommuting, but all offers will be considered. Contact eric at fnords.org. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream CallerID
Is it possible to send the incoming PSTN caller ID to a Grandstream Budge Tone-100 SIP phone? I’ve configured the extensions.conf file and the log is definitely showing the incoming caller ID information, but its not being sent to the SIP phone (assuming that’s possible). I’ve searched through the mailing list threads looking for an answer to this but came up empty handed. -Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GrandStream CallerID
Here you are: [general] language=de context=incoming; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) register => XXX:[EMAIL PROTECTED]/sipgatedial disallow=all; Disallow all codecs allow=gsm allow=alaw allow=ulaw [2200] ;grandstram bt-101 disallow=all allow=ulaw allow=alaw auth=plaintext type=friend callerid="Something" <2200> username=2200 dtmfmode=rfc2833 secret= host=dynamic context=out mailbox=2200 i hope this helps Am Wed, 26 May 2004 11:48:07 -0400 schrieb Steve Totaro <[EMAIL PROTECTED]>: need sip.conf do you have callerid=xxx specified for your phones? is the name the same as the extension? - Original Message - From: "shabanip" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, July 26, 2004 8:40 AM Subject: [Asterisk-Users] GrandStream CallerID I see my own number(or remote called num) instead of caller id on incoming calls on my BT-102. but on Xlite everyything is OK. I'm using * latest CVS. - shabanip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erstellt mit M2, Operas revolutionärem E-Mail-Modul: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GrandStream CallerID
I found where the problem comes from, i'd set the fromuser=exten in sip.conf it seems that it overwrites the received callerid. btw, thanks > need sip.conf > > do you have callerid=xxx specified for your phones? is the name the same > as > the extension? > > > - Original Message - > From: "shabanip" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Monday, July 26, 2004 8:40 AM > Subject: [Asterisk-Users] GrandStream CallerID > > >> I see my own number(or remote called num) instead of caller id on >> incoming >> calls on my BT-102. >> but on Xlite everyything is OK. I'm using * latest CVS. >> >> - shabanip >> >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GrandStream CallerID
need sip.conf do you have callerid=xxx specified for your phones? is the name the same as the extension? - Original Message - From: "shabanip" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, July 26, 2004 8:40 AM Subject: [Asterisk-Users] GrandStream CallerID > I see my own number(or remote called num) instead of caller id on incoming > calls on my BT-102. > but on Xlite everyything is OK. I'm using * latest CVS. > > - shabanip > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GrandStream CallerID
I see my own number(or remote called num) instead of caller id on incoming calls on my BT-102. but on Xlite everyything is OK. I'm using * latest CVS. - shabanip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
