Re: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P
Do you have a wait(2) before your dial(SIP/) ? You need to allow a full ring before you build your first sip packet. Jeremy Bogan wrote: Yeah I have callerid=asreceived in my zapata.conf still nothing unfortunately. I get that when the calling party has caller id blocked on their end. ___ 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] Passing CallerID to SIP phone from TDM400P
Yeah I have callerid=asreceived in my zapata.conf still nothing unfortunately. I get that when the calling party has caller id blocked on their end. -- jeremy bogan[ [EMAIL PROTECTED] ] segment publishing - design.develop.host ___ 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] Passing CallerID to SIP phone from TDM400P
--On Wednesday, October 13, 2004 16:04 +1000 James Bean [EMAIL PROTECTED] wrote: a) Ensure you actually have the callerid service provided to your line, this is usually an extra charge from telstra (AFAIK) Yep my analog handset on the line (not through asterisk) displays the callerid of the incoming call (just as a double check). I might be wrong here, but don'y you also need callerid=asreceived on the incoming Zap channel in zapata.conf as well? ___ 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] Passing CallerID to SIP phone from TDM400P
James - I have the same problem, and tried a some of the same ideas. No result. But at least we both know that a few people in Australia are using Asterisk! Later, PaulH -Original Message- From: James Bean [mailto:[EMAIL PROTECTED] Sent: Wednesday, 13 October 2004 4:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P Sorry, I explained this wrong. I am wanting the callerid of the incoming caller from my analogue line on the TDM400P to be passed TO the sip phone so the sip phone display shows the phone number of the incoming caler from the call on the TDM400P. It shows any callerid information from other sip phones or extension calls fine. I'm not sure, but try the following: a) Ensure you actually have the callerid service provided to your line, this is usually an extra charge from telstra (AFAIK) Yep my analog handset on the line (not through asterisk) displays the callerid of the incoming call (just as a double check). b) Add a wait(3) (maybe a 2 or 4 seconds is needed) before the noop Took it out to Wait(5), and made sure that the callerid was being displayed on my analog handset before the wait times out in asterisk to do the noop. Still no go. SIP handset still displays Asterisk on it when the call is patched through. c) Patch asterisk with this patch (I'm still waiting to be able to do this from a config file. This is what I use to allow asterisk to pass callerid *to* my analog FXS extensions. I assume it is the same for FXO lines. diff -ur asterisk/channels/chan_zap.c asterisk.mine/channels/chan_zap.c --- asterisk/channels/chan_zap.cWed Sep 22 18:24:18 2004 +++ asterisk.mine/channels/chan_zap.c Wed Sep 22 18:24:41 2004 @@ -89,7 +89,7 @@ /* #define ZAP_CHECK_HOOKSTATE */ /* Typically, how many rings before we should send Caller*ID */ -#define DEFAULT_CIDRINGS 1 +#define DEFAULT_CIDRINGS 2 #define CHANNEL_PSEUDO -12 Obviously after the last one, you need to re-compile and re-install asterisk, and then re-start asterisk. Regards, Adam Yes I had found this patch previous and it was already compiled into my current build, asterisk 1.0.1... Thanks for the reply though it did open my eyes to a few things. Unfortunately no callerid from the incoming analog line call on my TDM400P. James ___ 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 CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ 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] Passing CallerID to SIP phone from TDM400P
Yeah I have callerid=asreceived in my zapata.conf still nothing unfortunately. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Loftis Sent: Wednesday, 13 October 2004 4:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P --On Wednesday, October 13, 2004 16:04 +1000 James Bean [EMAIL PROTECTED] wrote: a) Ensure you actually have the callerid service provided to your line, this is usually an extra charge from telstra (AFAIK) Yep my analog handset on the line (not through asterisk) displays the callerid of the incoming call (just as a double check). I might be wrong here, but don'y you also need callerid=asreceived on the incoming Zap channel in zapata.conf as well? ___ 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] Passing CallerID to SIP phone from TDM400P
Its getting pretty well spread here with several ISP's/Telco's offering IAX connectivity for cheap calls. It's growing, I hope we can just sort out the callerid thing :-). Although I could name the line it comes in on so it doesn't just say asterisk when the call comes in. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, 13 October 2004 4:42 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P James - I have the same problem, and tried a some of the same ideas. No result. But at least we both know that a few people in Australia are using Asterisk! Later, PaulH -Original Message- From: James Bean [mailto:[EMAIL PROTECTED] Sent: Wednesday, 13 October 2004 4:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P Sorry, I explained this wrong. I am wanting the callerid of the incoming caller from my analogue line on the TDM400P to be passed TO the sip phone so the sip phone display shows the phone number of the incoming caler from the call on the TDM400P. It shows any callerid information from other sip phones or extension calls fine. I'm not sure, but try the following: a) Ensure you actually have the callerid service provided to your line, this is usually an extra charge from telstra (AFAIK) Yep my analog handset on the line (not through asterisk) displays the callerid of the incoming call (just as a double check). b) Add a wait(3) (maybe a 2 or 4 seconds is needed) before the noop Took it out to Wait(5), and made sure that the callerid was being displayed on my analog handset before the wait times out in asterisk to do the noop. Still no go. SIP handset still displays Asterisk on it when the call is patched through. c) Patch asterisk with this patch (I'm still waiting to be able to do this from a config file. This is what I use to allow asterisk to pass callerid *to* my analog FXS extensions. I assume it is the same for FXO lines. diff -ur asterisk/channels/chan_zap.c asterisk.mine/channels/chan_zap.c --- asterisk/channels/chan_zap.cWed Sep 22 18:24:18 2004 +++ asterisk.mine/channels/chan_zap.c Wed Sep 22 18:24:41 2004 @@ -89,7 +89,7 @@ /* #define ZAP_CHECK_HOOKSTATE */ /* Typically, how many rings before we should send Caller*ID */ -#define DEFAULT_CIDRINGS 1 +#define DEFAULT_CIDRINGS 2 #define CHANNEL_PSEUDO -12 Obviously after the last one, you need to re-compile and re-install asterisk, and then re-start asterisk. Regards, Adam Yes I had found this patch previous and it was already compiled into my current build, asterisk 1.0.1... Thanks for the reply though it did open my eyes to a few things. Unfortunately no callerid from the incoming analog line call on my TDM400P. James ___ 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 CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ 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] Passing CallerID to SIP phone from TDM400P
The important part is to keep the discussion on the list, so other people can benefit from our work! PaulH -Original Message- From: James Bean [mailto:[EMAIL PROTECTED] Sent: Wednesday, 13 October 2004 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P Its getting pretty well spread here with several ISP's/Telco's offering IAX connectivity for cheap calls. It's growing, I hope we can just sort out the callerid thing :-). Although I could name the line it comes in on so it doesn't just say asterisk when the call comes in. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, 13 October 2004 4:42 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P James - I have the same problem, and tried a some of the same ideas. No result. But at least we both know that a few people in Australia are using Asterisk! Later, PaulH -Original Message- From: James Bean [mailto:[EMAIL PROTECTED] Sent: Wednesday, 13 October 2004 4:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P Sorry, I explained this wrong. I am wanting the callerid of the incoming caller from my analogue line on the TDM400P to be passed TO the sip phone so the sip phone display shows the phone number of the incoming caler from the call on the TDM400P. It shows any callerid information from other sip phones or extension calls fine. I'm not sure, but try the following: a) Ensure you actually have the callerid service provided to your line, this is usually an extra charge from telstra (AFAIK) Yep my analog handset on the line (not through asterisk) displays the callerid of the incoming call (just as a double check). b) Add a wait(3) (maybe a 2 or 4 seconds is needed) before the noop Took it out to Wait(5), and made sure that the callerid was being displayed on my analog handset before the wait times out in asterisk to do the noop. Still no go. SIP handset still displays Asterisk on it when the call is patched through. c) Patch asterisk with this patch (I'm still waiting to be able to do this from a config file. This is what I use to allow asterisk to pass callerid *to* my analog FXS extensions. I assume it is the same for FXO lines. diff -ur asterisk/channels/chan_zap.c asterisk.mine/channels/chan_zap.c --- asterisk/channels/chan_zap.cWed Sep 22 18:24:18 2004 +++ asterisk.mine/channels/chan_zap.c Wed Sep 22 18:24:41 2004 @@ -89,7 +89,7 @@ /* #define ZAP_CHECK_HOOKSTATE */ /* Typically, how many rings before we should send Caller*ID */ -#define DEFAULT_CIDRINGS 1 +#define DEFAULT_CIDRINGS 2 #define CHANNEL_PSEUDO -12 Obviously after the last one, you need to re-compile and re-install asterisk, and then re-start asterisk. Regards, Adam Yes I had found this patch previous and it was already compiled into my current build, asterisk 1.0.1... Thanks for the reply though it did open my eyes to a few things. Unfortunately no callerid from the incoming analog line call on my TDM400P. James ___ 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 CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ 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 CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination
Re: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P
If the extension is a SIP Phone, it's up to the SIP Phone to pass the
CallerID information. Some ATAs allow you to configure how's the
Caller_ID being transmitted (like Cisco ATA-186). Others don't.
if you call from the console, the Caller ID information will say
'asterisk'. from your phones, it won't.
If the call originates, for example, from a SIP endpoint (phone, etc).
it uses the callerid defined on sip.conf.
In your example, take the double quotes off (that seems to work in my
case):
[bt-karen]
type=friend
secret=password removed
host=dynamic
callerid=Karen 691
defaultip=192.168.69.251
dtmfmode=inband
mailbox=691
That would be what I would do.
On Oct 13, 2004, at 12:38 AM, James Bean wrote:
Hi,
Sorry, newbie, I want to pass the incoming callerid information
through to my sip phone but when an incoming call gets passed through
it says asterisk on the display instead of the number.
Being in australia callerid information is passed through on the
second ring not the first, (hence my noop command doesn't currently
work)
James
--
/etc/asterisk/extensions.conf
[pstn]
exten = s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a
comment in the CLI for info.
exten = s,2,Dial(SIP/snom-james,45,t)
exten = s,3,Hangup
;exten = s,3,VoiceMail(u100) ;Whatever box you want.
[internal]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone
include = sip
[sip]
exten = 690,1,Dial(SIP/snom-james,30,tr)
exten = 690,2,voicemail2,u900
exten = 690,102,voicemail2,b900
exten = 691,1,Dial(SIP/bt-karen,30,tr)
exten = 691,2,voicemail2,u901
exten = 691,102,voicemail,b901
/etc/asterisk/sip.conf
[general]
port = 5060
bindaddr = 192.168.69.1
context = sip
disallow = gsm
allow = alaw
disallow = ulaw
srvlookup=no
[snom-james]
type=friend
secret=password removed
host=dynamic
callerid=James 690
defaultip=192.168.69.250
dtmfmode=inband
mailbox=690
[bt-karen]
type=friend
secret=password removed
host=dynamic
callerid=Karen 691
defaultip=192.168.69.251
dtmfmode=inband
mailbox=691
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RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P
Sorry, I explained this wrong.
I am wanting the callerid of the incoming caller from my analogue line on the TDM400P
to be passed TO the sip phone so the sip phone display shows the phone number of the
incoming caler from the call on the TDM400P.
It shows any callerid information from other sip phones or extension calls fine.
James
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Emilio Panighetti
Sent: Wednesday, 13 October 2004 2:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P
If the extension is a SIP Phone, it's up to the SIP Phone to pass the CallerID
information. Some ATAs allow you to configure how's the Caller_ID being transmitted
(like Cisco ATA-186). Others don't.
if you call from the console, the Caller ID information will say 'asterisk'. from your
phones, it won't.
If the call originates, for example, from a SIP endpoint (phone, etc).
it uses the callerid defined on sip.conf.
In your example, take the double quotes off (that seems to work in my
case):
[bt-karen]
type=friend
secret=password removed
host=dynamic
callerid=Karen 691
defaultip=192.168.69.251
dtmfmode=inband
mailbox=691
That would be what I would do.
On Oct 13, 2004, at 12:38 AM, James Bean wrote:
Hi,
Sorry, newbie, I want to pass the incoming callerid information
through to my sip phone but when an incoming call gets passed through
it says asterisk on the display instead of the number.
Being in australia callerid information is passed through on the
second ring not the first, (hence my noop command doesn't currently
work)
James
--
/etc/asterisk/extensions.conf
[pstn]
exten = s,1,NoOp(Comment Only: Call from ${CALLERIDNUM}) ; Just put a
comment in the CLI for info.
exten = s,2,Dial(SIP/snom-james,45,t) exten = s,3,Hangup ;exten
= s,3,VoiceMail(u100) ;Whatever box you want.
[internal]
exten = i,1,Playback(invalid)
exten = i,2,Hangup
exten = t,1,Hangup
exten = 099,1,Echo ;simple echo test when you dial 099 on your
phone
include = sip
[sip]
exten = 690,1,Dial(SIP/snom-james,30,tr) exten =
690,2,voicemail2,u900 exten = 690,102,voicemail2,b900
exten = 691,1,Dial(SIP/bt-karen,30,tr) exten =
691,2,voicemail2,u901 exten = 691,102,voicemail,b901
/etc/asterisk/sip.conf
[general]
port = 5060
bindaddr = 192.168.69.1
context = sip
disallow = gsm
allow = alaw
disallow = ulaw
srvlookup=no
[snom-james]
type=friend
secret=password removed
host=dynamic
callerid=James 690
defaultip=192.168.69.250
dtmfmode=inband
mailbox=690
[bt-karen]
type=friend
secret=password removed
host=dynamic
callerid=Karen 691
defaultip=192.168.69.251
dtmfmode=inband
mailbox=691
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RE: [Asterisk-Users] Passing CallerID to SIP phone from TDM400P
On Wed, 2004-10-13 at 15:06, James Bean wrote: Sorry, I explained this wrong. I am wanting the callerid of the incoming caller from my analogue line on the TDM400P to be passed TO the sip phone so the sip phone display shows the phone number of the incoming caler from the call on the TDM400P. It shows any callerid information from other sip phones or extension calls fine. I'm not sure, but try the following: a) Ensure you actually have the callerid service provided to your line, this is usually an extra charge from telstra (AFAIK) b) Add a wait(3) (maybe a 2 or 4 seconds is needed) before the noop c) Patch asterisk with this patch (I'm still waiting to be able to do this from a config file. This is what I use to allow asterisk to pass callerid *to* my analog FXS extensions. I assume it is the same for FXO lines. diff -ur asterisk/channels/chan_zap.c asterisk.mine/channels/chan_zap.c --- asterisk/channels/chan_zap.cWed Sep 22 18:24:18 2004 +++ asterisk.mine/channels/chan_zap.c Wed Sep 22 18:24:41 2004 @@ -89,7 +89,7 @@ /* #define ZAP_CHECK_HOOKSTATE */ /* Typically, how many rings before we should send Caller*ID */ -#define DEFAULT_CIDRINGS 1 +#define DEFAULT_CIDRINGS 2 #define CHANNEL_PSEUDO -12 Obviously after the last one, you need to re-compile and re-install asterisk, and then re-start asterisk. Regards, Adam ___ 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] Passing CallerID to SIP phone from TDM400P
Sorry, I explained this wrong. I am wanting the callerid of the incoming caller from my analogue line on the TDM400P to be passed TO the sip phone so the sip phone display shows the phone number of the incoming caler from the call on the TDM400P. It shows any callerid information from other sip phones or extension calls fine. I'm not sure, but try the following: a) Ensure you actually have the callerid service provided to your line, this is usually an extra charge from telstra (AFAIK) Yep my analog handset on the line (not through asterisk) displays the callerid of the incoming call (just as a double check). b) Add a wait(3) (maybe a 2 or 4 seconds is needed) before the noop Took it out to Wait(5), and made sure that the callerid was being displayed on my analog handset before the wait times out in asterisk to do the noop. Still no go. SIP handset still displays Asterisk on it when the call is patched through. c) Patch asterisk with this patch (I'm still waiting to be able to do this from a config file. This is what I use to allow asterisk to pass callerid *to* my analog FXS extensions. I assume it is the same for FXO lines. diff -ur asterisk/channels/chan_zap.c asterisk.mine/channels/chan_zap.c --- asterisk/channels/chan_zap.cWed Sep 22 18:24:18 2004 +++ asterisk.mine/channels/chan_zap.c Wed Sep 22 18:24:41 2004 @@ -89,7 +89,7 @@ /* #define ZAP_CHECK_HOOKSTATE */ /* Typically, how many rings before we should send Caller*ID */ -#define DEFAULT_CIDRINGS 1 +#define DEFAULT_CIDRINGS 2 #define CHANNEL_PSEUDO -12 Obviously after the last one, you need to re-compile and re-install asterisk, and then re-start asterisk. Regards, Adam Yes I had found this patch previous and it was already compiled into my current build, asterisk 1.0.1... Thanks for the reply though it did open my eyes to a few things. Unfortunately no callerid from the incoming analog line call on my TDM400P. James ___ 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
