OK here it goes.. Caller ID is two parts or actually three: Part 1 Number only Part 2 Number + Name Part 3 Whole lotta stuff (also known as ADSI) Here is the US, I cannot speak for other countries. When party A places a call to Party B. Party A's Telco picks up the number, either from a table on the switch or passed from the PRI from Party A. Then on the far side (Party B's Telco) the Telco does a lookup in the LIDB (Line Information Data Base) and associates a name with a number. This information is then passed as Part II CLID. I have simplified the process, leaving out many processes along the way but it should give some insight as to how the Name actually shows up on the other end. Most Telcos do not receive the Name as part of the data in the call through the tandems b/w Telcos, they opt rather to do the lookup in the LIDB themselves. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F Sent: Sunday, January 09, 2005 6:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Little confused about Caller ID When calling to the PSTN (outside VOIP or *) then you will not be able to supply the name of callerID even if you have a PRI. The only thing you can provide is the number and the receiving switch of the call is the one responsibble for attaching a name to the phone number thru SS7. If you have a SS7 switch then you could in theory attach the name (I have never tried it, but that's what I was told). Hope this helps. On Sat, 8 Jan 2005 23:23:45 -0500 (EST), Samuel T. Cossette <digium@muel.org> wrote:> Hi, > > I've got the Caller ID name and number working with the application > SetCIDNumber and SetCIDName. > > [...] > exten => s,3,SetCIDNumber(4183289901) > exten => s,4,SetCIDName(Frank Black) > exten => s,5,Dial(IAX2/prov01/${DEST}) > [...] > > You can also use SetCallerID(Frank Black <4183289901>), but no successfor> me... > > bye, > > Samuel T. Cossette > samuel@levinux.org, 1.418.8o2.784o > << Well, that's for me to know and you to find out. >> Jeffrey, BlueVelvet> > > Hi Everybody, > > Sure this has been covered a million times on wiki, but couldn'tfind> > an > > exact answer to my question. I am using * to dial out to peoplesphones> > to > > give them alerts of different things. Problem is that the onlyCaller ID> > I > > can get working is the telephone number. I am unable to display aname> > along with the number. Thinking maybe its the phone receiving thecall, I> > tried my cellphone and my house phone and I can only get the numberto> > display. If I leave the number portion out, Caller ID shows > > "Unavailable". > > Is there a simple way to get a Caller name setup? I've triedexamples on> > Wiki as well but I couldn't get them to work. > > > > > > ***** extensions.conf ********* > > [general] > > static=yes > > writeprotect=no > > > > [globals] > > CONSOLE=Console/dsp ; Console interface for demo > > TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) > > > > [sports] > > exten => s,1,ResponseTimeout,5 > > exten => s,2,Answer > > exten => s,3,Wait(1) > > exten => s,4,Playback(sports/gafanaSports) > > exten => s,5,Goto(2000,2) > > exten => 2000,1,wait(1) > > exten => 2000,2,Background(sports/teams/theLosAngelesLakersU) > > exten => t,1,Playback(goodbye) > > exten => t,2,Hangup > > > > ******** sip.conf *********** > > [general] > > context=default > > port=5060 > > srvlookup=yes > > allow=ulaw > > register => [id]:[pw]@[host] > > [gafana] > > type=peer > > secret=[secret] > > username=[username] > > host=[hostname] > > > > > > ******* test.call file ********** > > Channel: SIP/[myNumber]@gafana > > CallerID: [My Number] > > MaxRetries: 0 > > RetryTime: 300 > > WaitTime: 45 > > Context: sports > > Extension: s > > Priority: 1 > > > > > > What do I need to add to be able to send a name as well and not justa> > number? > > > > Gabe > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > 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 > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Alexander Lopez wrote:> > Most Telcos do not receive the Name as part of the data in the call > through the tandems b/w Telcos, they opt rather to do the lookup in the > LIDB themselves. >Just for the sake of completeness: "most telcos do not" would imply that "some telcos *do*." You also say they "opt" to to the lookup. Does this mean that the name is actually sent along, but ignored by most endpoint switches, or is there something else going on beyond that? Thx. B.
Is the TCAP DB part of the LIDB collective (no Borg pun intended)?? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Chandler Sent: Sunday, January 09, 2005 6:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; C F Subject: Re: [Asterisk-Users] Little confused about Caller ID Caller Name is stored in a SCP. It is a TCAP transaction. The receiving switch via SS7 recieves the calling party number in the ISUP message of the SS7 datastream. It is normally in the IAM mesasge. Then a TCAP CNAME query is launched from the called switch thru the STP's to a SCP which has the calling name database. The TCAP query returns back to the launching switch the caller name. LIDB is for operator services etc. CNAME is a TCAP database lookup, much like 800 number translations. Tom C. ----- Original Message ----- From: "Alexander Lopez" <alex.lopez@opsys.com> To: "C F" <shmaltz@gmail.com>; "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Sunday, January 09, 2005 5:30 PM Subject: RE: [Asterisk-Users] Little confused about Caller ID> OK here it goes.. > > Caller ID is two parts or actually three: > > Part 1 Number only > Part 2 Number + Name > Part 3 Whole lotta stuff (also known as ADSI) > > > Here is the US, I cannot speak for other countries. > > When party A places a call to Party B. Party A's Telco picks up the > number, either from a table on the switch or passed from the PRI from > Party A. Then on the far side (Party B's Telco) the Telco does alookup> in the LIDB (Line Information Data Base) and associates a name with a > number. This information is then passed as Part II CLID. > > I have simplified the process, leaving out many processes along theway> but it should give some insight as to how the Name actually shows upon> the other end. > > Most Telcos do not receive the Name as part of the data in the call > through the tandems b/w Telcos, they opt rather to do the lookup inthe> LIDB themselves. > > > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F > Sent: Sunday, January 09, 2005 6:16 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Little confused about Caller ID > > When calling to the PSTN (outside VOIP or *) then you will not be able > to supply the name of callerID even if you have a PRI. The only thing > you can provide is the number and the receiving switch of the call is > the one responsibble for attaching a name to the phone number thru > SS7. If you have a SS7 switch then you could in theory attach the name > (I have never tried it, but that's what I was told). > Hope this helps. > > > > On Sat, 8 Jan 2005 23:23:45 -0500 (EST), Samuel T. Cossette > <digium@muel.org> wrote: > > Hi, > > > > I've got the Caller ID name and number working with the application > > SetCIDNumber and SetCIDName. > > > > [...] > > exten => s,3,SetCIDNumber(4183289901) > > exten => s,4,SetCIDName(Frank Black) > > exten => s,5,Dial(IAX2/prov01/${DEST}) > > [...] > > > > You can also use SetCallerID(Frank Black <4183289901>), but nosuccess> for > > me... > > > > bye, > > > > Samuel T. Cossette > > samuel@levinux.org, 1.418.8o2.784o > > << Well, that's for me to know and you to find out. >> Jeffrey, Blue > Velvet > > > > > Hi Everybody, > > > Sure this has been covered a million times on wiki, butcouldn't> find > > > an > > > exact answer to my question. I am using * to dial out to peoples > phones > > > to > > > give them alerts of different things. Problem is that the only > Caller ID > > > I > > > can get working is the telephone number. I am unable to display a > name > > > along with the number. Thinking maybe its the phone receiving the > call, I > > > tried my cellphone and my house phone and I can only get thenumber> to > > > display. If I leave the number portion out, Caller ID shows > > > "Unavailable". > > > Is there a simple way to get a Caller name setup? I've tried > examples on > > > Wiki as well but I couldn't get them to work. > > > > > > > > > ***** extensions.conf ********* > > > [general] > > > static=yes > > > writeprotect=no > > > > > > [globals] > > > CONSOLE=Console/dsp ; Console interface for demo > > > TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) > > > > > > [sports] > > > exten => s,1,ResponseTimeout,5 > > > exten => s,2,Answer > > > exten => s,3,Wait(1) > > > exten => s,4,Playback(sports/gafanaSports) > > > exten => s,5,Goto(2000,2) > > > exten => 2000,1,wait(1) > > > exten => 2000,2,Background(sports/teams/theLosAngelesLakersU) > > > exten => t,1,Playback(goodbye) > > > exten => t,2,Hangup > > > > > > ******** sip.conf *********** > > > [general] > > > context=default > > > port=5060 > > > srvlookup=yes > > > allow=ulaw > > > register => [id]:[pw]@[host] > > > [gafana] > > > type=peer > > > secret=[secret] > > > username=[username] > > > host=[hostname] > > > > > > > > > ******* test.call file ********** > > > Channel: SIP/[myNumber]@gafana > > > CallerID: [My Number] > > > MaxRetries: 0 > > > RetryTime: 300 > > > WaitTime: 45 > > > Context: sports > > > Extension: s > > > Priority: 1 > > > > > > > > > What do I need to add to be able to send a name as well and notjust> a > > > number? > > > > > > Gabe > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > 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 > > Asterisk-Users@lists.digium.com > > 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 > Asterisk-Users@lists.digium.com > 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 > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ----------------------------------------------- > Scanned by Bayou Internet for all known viruses. > http://www.bayou.com > >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Caller Name is stored in a SCP. It is a TCAP transaction. The receiving switch via SS7 recieves the calling party number in the ISUP message of the SS7 datastream. It is normally in the IAM mesasge. Then a TCAP CNAME query is launched from the called switch thru the STP's to a SCP which has the calling name database. The TCAP query returns back to the launching switch the caller name. LIDB is for operator services etc. CNAME is a TCAP database lookup, much like 800 number translations. Tom C. ----- Original Message ----- From: "Alexander Lopez" <alex.lopez@opsys.com> To: "C F" <shmaltz@gmail.com>; "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Sunday, January 09, 2005 5:30 PM Subject: RE: [Asterisk-Users] Little confused about Caller ID> OK here it goes.. > > Caller ID is two parts or actually three: > > Part 1 Number only > Part 2 Number + Name > Part 3 Whole lotta stuff (also known as ADSI) > > > Here is the US, I cannot speak for other countries. > > When party A places a call to Party B. Party A's Telco picks up the > number, either from a table on the switch or passed from the PRI from > Party A. Then on the far side (Party B's Telco) the Telco does a lookup > in the LIDB (Line Information Data Base) and associates a name with a > number. This information is then passed as Part II CLID. > > I have simplified the process, leaving out many processes along the way > but it should give some insight as to how the Name actually shows up on > the other end. > > Most Telcos do not receive the Name as part of the data in the call > through the tandems b/w Telcos, they opt rather to do the lookup in the > LIDB themselves. > > > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F > Sent: Sunday, January 09, 2005 6:16 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Little confused about Caller ID > > When calling to the PSTN (outside VOIP or *) then you will not be able > to supply the name of callerID even if you have a PRI. The only thing > you can provide is the number and the receiving switch of the call is > the one responsibble for attaching a name to the phone number thru > SS7. If you have a SS7 switch then you could in theory attach the name > (I have never tried it, but that's what I was told). > Hope this helps. > > > > On Sat, 8 Jan 2005 23:23:45 -0500 (EST), Samuel T. Cossette > <digium@muel.org> wrote: > > Hi, > > > > I've got the Caller ID name and number working with the application > > SetCIDNumber and SetCIDName. > > > > [...] > > exten => s,3,SetCIDNumber(4183289901) > > exten => s,4,SetCIDName(Frank Black) > > exten => s,5,Dial(IAX2/prov01/${DEST}) > > [...] > > > > You can also use SetCallerID(Frank Black <4183289901>), but no success > for > > me... > > > > bye, > > > > Samuel T. Cossette > > samuel@levinux.org, 1.418.8o2.784o > > << Well, that's for me to know and you to find out. >> Jeffrey, Blue > Velvet > > > > > Hi Everybody, > > > Sure this has been covered a million times on wiki, but couldn't > find > > > an > > > exact answer to my question. I am using * to dial out to peoples > phones > > > to > > > give them alerts of different things. Problem is that the only > Caller ID > > > I > > > can get working is the telephone number. I am unable to display a > name > > > along with the number. Thinking maybe its the phone receiving the > call, I > > > tried my cellphone and my house phone and I can only get the number > to > > > display. If I leave the number portion out, Caller ID shows > > > "Unavailable". > > > Is there a simple way to get a Caller name setup? I've tried > examples on > > > Wiki as well but I couldn't get them to work. > > > > > > > > > ***** extensions.conf ********* > > > [general] > > > static=yes > > > writeprotect=no > > > > > > [globals] > > > CONSOLE=Console/dsp ; Console interface for demo > > > TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) > > > > > > [sports] > > > exten => s,1,ResponseTimeout,5 > > > exten => s,2,Answer > > > exten => s,3,Wait(1) > > > exten => s,4,Playback(sports/gafanaSports) > > > exten => s,5,Goto(2000,2) > > > exten => 2000,1,wait(1) > > > exten => 2000,2,Background(sports/teams/theLosAngelesLakersU) > > > exten => t,1,Playback(goodbye) > > > exten => t,2,Hangup > > > > > > ******** sip.conf *********** > > > [general] > > > context=default > > > port=5060 > > > srvlookup=yes > > > allow=ulaw > > > register => [id]:[pw]@[host] > > > [gafana] > > > type=peer > > > secret=[secret] > > > username=[username] > > > host=[hostname] > > > > > > > > > ******* test.call file ********** > > > Channel: SIP/[myNumber]@gafana > > > CallerID: [My Number] > > > MaxRetries: 0 > > > RetryTime: 300 > > > WaitTime: 45 > > > Context: sports > > > Extension: s > > > Priority: 1 > > > > > > > > > What do I need to add to be able to send a name as well and not just > a > > > number? > > > > > > Gabe > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > 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 > > Asterisk-Users@lists.digium.com > > 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 > Asterisk-Users@lists.digium.com > 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 > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ----------------------------------------------- > Scanned by Bayou Internet for all known viruses. > http://www.bayou.com > >
In my six years of SS7 work I have never seen the calling name generated by the calling switch and passed via the SS7 network. Normally and of all the installations that I seen, it is done by the called switch via a TCAP query to a SCP database. Tom c. ----- Original Message ----- From: "Brian Capouch" <brianc@palaver.net> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Sunday, January 09, 2005 5:32 PM Subject: Re: [Asterisk-Users] Little confused about Caller ID> Alexander Lopez wrote: > > > > Most Telcos do not receive the Name as part of the data in the call > > through the tandems b/w Telcos, they opt rather to do the lookup in the > > LIDB themselves. > > > > Just for the sake of completeness: "most telcos do not" would imply that > "some telcos *do*." > > You also say they "opt" to to the lookup. > > Does this mean that the name is actually sent along, but ignored by most > endpoint switches, or is there something else going on beyond that? > > Thx. > > B. > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ----------------------------------------------- > Scanned by Bayou Internet for all known viruses. > http://www.bayou.com >
Thanks. I was always under the impression that they were all separate tables in the same DB and that they were collectively called 'The LIDB!!' For my and the others here could you describe the function of the different DBs? I now understand the CNAME, I thought I knew the LIBD, I can guess on the LNP, and 800, but what about the AIN??? BTW. I was under the impression the fields had been added to the LIDB to handle the Do Not Call list. Can anyone confirm or deny?? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Chandler Sent: Sunday, January 09, 2005 6:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Little confused about Caller ID TCAP is a transaction application. The CNAME, LIDB,800,.LNP and AIN database COULD be in the same SCP, but in most cases it is not. LIDB database are used for calling card, operator services, etc. These are all seperate databases stored for use in an SCP connected to STP's. So is there a relationship between CNAME and LIBD, no. Tom C. ----- Original Message ----- From: "Alexander Lopez" <alex.lopez@opsys.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Sunday, January 09, 2005 5:44 PM Subject: RE: [Asterisk-Users] Little confused about Caller ID> Is the TCAP DB part of the LIDB collective (no Borg pun intended)?? > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom > Chandler > Sent: Sunday, January 09, 2005 6:45 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion; C F > Subject: Re: [Asterisk-Users] Little confused about Caller ID > > Caller Name is stored in a SCP. It is a TCAP transaction. The > receiving > switch via SS7 recieves > the calling party number in the ISUP message of the SS7 datastream.It> is > normally in the IAM mesasge. Then a TCAP CNAME query is launched from > the > called switch thru > the STP's to a SCP which has the calling name database. The TCAPquery> returns back to the launching > switch the caller name. LIDB is for operator services etc. CNAME is a > TCAP > database lookup, much > like 800 number translations. > > Tom C. > > ----- Original Message ----- > From: "Alexander Lopez" <alex.lopez@opsys.com> > To: "C F" <shmaltz@gmail.com>; "Asterisk Users Mailing List - > Non-Commercial > Discussion" <asterisk-users@lists.digium.com> > Sent: Sunday, January 09, 2005 5:30 PM > Subject: RE: [Asterisk-Users] Little confused about Caller ID > > > > OK here it goes.. > > > > Caller ID is two parts or actually three: > > > > Part 1 Number only > > Part 2 Number + Name > > Part 3 Whole lotta stuff (also known as ADSI) > > > > > > Here is the US, I cannot speak for other countries. > > > > When party A places a call to Party B. Party A's Telco picks up the > > number, either from a table on the switch or passed from the PRIfrom> > Party A. Then on the far side (Party B's Telco) the Telco does a > lookup > > in the LIDB (Line Information Data Base) and associates a name witha> > number. This information is then passed as Part II CLID. > > > > I have simplified the process, leaving out many processes along the > way > > but it should give some insight as to how the Name actually shows up > on > > the other end. > > > > Most Telcos do not receive the Name as part of the data in the call > > through the tandems b/w Telcos, they opt rather to do the lookup in > the > > LIDB themselves. > > > > > > > > > > -----Original Message----- > > From: asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F > > Sent: Sunday, January 09, 2005 6:16 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] Little confused about Caller ID > > > > When calling to the PSTN (outside VOIP or *) then you will not beable> > to supply the name of callerID even if you have a PRI. The onlything> > you can provide is the number and the receiving switch of the callis> > the one responsibble for attaching a name to the phone number thru > > SS7. If you have a SS7 switch then you could in theory attach thename> > (I have never tried it, but that's what I was told). > > Hope this helps. > > > > > > > > On Sat, 8 Jan 2005 23:23:45 -0500 (EST), Samuel T. Cossette > > <digium@muel.org> wrote: > > > Hi, > > > > > > I've got the Caller ID name and number working with theapplication> > > SetCIDNumber and SetCIDName. > > > > > > [...] > > > exten => s,3,SetCIDNumber(4183289901) > > > exten => s,4,SetCIDName(Frank Black) > > > exten => s,5,Dial(IAX2/prov01/${DEST}) > > > [...] > > > > > > You can also use SetCallerID(Frank Black <4183289901>), but no > success > > for > > > me... > > > > > > bye, > > > > > > Samuel T. Cossette > > > samuel@levinux.org, 1.418.8o2.784o > > > << Well, that's for me to know and you to find out. >> Jeffrey,Blue> > Velvet > > > > > > > Hi Everybody, > > > > Sure this has been covered a million times on wiki, but > couldn't > > find > > > > an > > > > exact answer to my question. I am using * to dial out topeoples> > phones > > > > to > > > > give them alerts of different things. Problem is that the only > > Caller ID > > > > I > > > > can get working is the telephone number. I am unable to displaya> > name > > > > along with the number. Thinking maybe its the phone receivingthe> > call, I > > > > tried my cellphone and my house phone and I can only get the > number > > to > > > > display. If I leave the number portion out, Caller ID shows > > > > "Unavailable". > > > > Is there a simple way to get a Caller name setup? I've tried > > examples on > > > > Wiki as well but I couldn't get them to work. > > > > > > > > > > > > ***** extensions.conf ********* > > > > [general] > > > > static=yes > > > > writeprotect=no > > > > > > > > [globals] > > > > CONSOLE=Console/dsp ; Console interface for demo > > > > TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) > > > > > > > > [sports] > > > > exten => s,1,ResponseTimeout,5 > > > > exten => s,2,Answer > > > > exten => s,3,Wait(1) > > > > exten => s,4,Playback(sports/gafanaSports) > > > > exten => s,5,Goto(2000,2) > > > > exten => 2000,1,wait(1) > > > > exten => 2000,2,Background(sports/teams/theLosAngelesLakersU) > > > > exten => t,1,Playback(goodbye) > > > > exten => t,2,Hangup > > > > > > > > ******** sip.conf *********** > > > > [general] > > > > context=default > > > > port=5060 > > > > srvlookup=yes > > > > allow=ulaw > > > > register => [id]:[pw]@[host] > > > > [gafana] > > > > type=peer > > > > secret=[secret] > > > > username=[username] > > > > host=[hostname] > > > > > > > > > > > > ******* test.call file ********** > > > > Channel: SIP/[myNumber]@gafana > > > > CallerID: [My Number] > > > > MaxRetries: 0 > > > > RetryTime: 300 > > > > WaitTime: 45 > > > > Context: sports > > > > Extension: s > > > > Priority: 1 > > > > > > > > > > > > What do I need to add to be able to send a name as well and not > just > > a > > > > number? > > > > > > > > Gabe > > > > > > > > _______________________________________________ > > > > Asterisk-Users mailing list > > > > Asterisk-Users@lists.digium.com > > > > 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 > > > Asterisk-Users@lists.digium.com > > > 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 > > Asterisk-Users@lists.digium.com > > 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 > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ----------------------------------------------- > > Scanned by Bayou Internet for all known viruses. > > http://www.bayou.com > > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ----------------------------------------------- > Scanned by Bayou Internet for all known viruses. > http://www.bayou.com > >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users