James Bean
2004-Oct-12 21:38 UTC
[Asterisk-Users] Passing CallerID to SIP phone from TDM400P
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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Emilio Panighetti
2004-Oct-12 21:38 UTC
[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=691That 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 > _______________________________________________ > 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
James Bean
2004-Oct-12 22:06 UTC
[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: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] 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=691That 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 > _______________________________________________ > 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
James Bean
2004-Oct-12 23:04 UTC
[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 analogueline>> 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 noopTook 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 dothis 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.c Wed 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-installasterisk, and then re-start asterisk.> >Regards, >AdamYes 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
That worked. Thx Eric. ----- Original Message ----- From: "Eric Wieling" <eric@fnords.org> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Wednesday, October 13, 2004 7:35 PM Subject: Re: [Asterisk-Users] SIP accepts all calls> spkao wrote: > > >Wonder if anyone has experienced this. I setup the SIP on * and I foundthat> >it will accept all calls does not matter if the username or secretmatches> >any > >client definition in sip.conf or not. > > > > > I thought that was fixed months ago.. You are either running an older > Asterisk or you have insecure=very in sip.conf. What I did to work > around the problem is put context=INVALID in [general] in sip.conf and > then put a context= line with the right context in each peer/usr/friend > entry in sip.conf. >---------------------------------------------------------------------------- ----> _______________________________________________ > 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
James Bean
2004-Oct-13 01:54 UTC
[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: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] 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 <james@hdcs.com.au> 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 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
James Bean
2004-Oct-13 01:56 UTC
[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: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] 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:james@hdcs.com.au] 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 analogueline>> 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 noopTook 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 dothis 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.c Wed 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-installasterisk, and then re-start asterisk.> >Regards, >AdamYes 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 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 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 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
Hello all, I am tring to change the default language in Asterisk, exactly for the Voicemail messages. I trying with the option Language=fr in the voicemail.conf global section, without success. I trying with the Setlanguage(fr) in the extensions.conf global section, but without success too. How could I change the default Languaje for Voicemail? I have got a /var/lib/asterisk/sounds/fr/ with all the sounds, i have a letter and diggits directory too. Any clue will be appreciated. Regards. Ismael Gil.
El 13/10/2004, a las 12:48, ismaelg escribi?:> How could I change the default Languaje for Voicemail? > > I have got a /var/lib/asterisk/sounds/fr/ with all the sounds, i have > a letter and diggits directory too. > > Any clue will be appreciated. >Mine is running fine, try it. exten => 207,1,Dial(SIP/007@192.168.0.100,10,Ttr) exten => 207,2,SetLanguage,fr exten => 207,3,Voicemail(${EXTEN}) exten => 207,4,Hangup Adri? Vidal mailto:adriavidal@telefonica.net
spkao wrote:>Wonder if anyone has experienced this. I setup the SIP on * and I found that >it will accept all calls does not matter if the username or secret matches >any >client definition in sip.conf or not. > >I thought that was fixed months ago.. You are either running an older Asterisk or you have insecure=very in sip.conf. What I did to work around the problem is put context=INVALID in [general] in sip.conf and then put a context= line with the right context in each peer/usr/friend entry in sip.conf. -------------- next part -------------- A non-text attachment was scrubbed... Name: eric.vcf Type: text/x-vcard Size: 146 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041013/1bdfb51a/eric.vcf
I am having the same problem. I noticed it today when I removed the section in sip.conf for my grandstream. I was testing the res_mysql_config realtime driver, so I put the grandstream config into the sip table of my database. * answered the calls into a demo ivr application as usually. So I removed the row in the sip table for the phone and called again. The call was still answered. I am using cvs-head that I compiled today. Is there a bugtrakker on this issue? I assume that * should NOT answer an inbound call from a sip client not listed in sip.conf or in the sip table (when using res_mysql_config)? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of spkao Sent: Wednesday, October 13, 2004 4:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP accepts all calls That worked. Thx Eric. ----- Original Message ----- From: "Eric Wieling" <eric@fnords.org> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Wednesday, October 13, 2004 7:35 PM Subject: Re: [Asterisk-Users] SIP accepts all calls> spkao wrote: > > >Wonder if anyone has experienced this. I setup the SIP on * and I foundthat> >it will accept all calls does not matter if the username or secretmatches> >any > >client definition in sip.conf or not. > > > > > I thought that was fixed months ago.. You are either running an older > Asterisk or you have insecure=very in sip.conf. What I did to work > around the problem is put context=INVALID in [general] in sip.conf and > then put a context= line with the right context in each peer/usr/friend > entry in sip.conf. >---------------------------------------------------------------------------- ----> _______________________________________________ > 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
Colin Haxton
2004-Oct-13 21:10 UTC
[Asterisk-Users] Passing CallerID to SIP phone from TDM400p
Hi James, I had a problem with caller ID where the id was being detected by * but was not being passed to the phone on a TDM400. What it turned out to be was the ringcadance. New Zealand, and Australia have a ringcadance of 400,200,400,2000 the TDM tried to send the caller-id after the first ring, this was only 200 and too small. I changed the ring to 400,2000,400,200. Sounds fine as you don't notice the slight half step at the beginning and caller-id has a good gap to be sent in. I don't know if this is affecting you but annoyed me for a while. Cheers, Colin -- ------------------------ Colin Haxton Arena Consulting Limited www.arena.co.nz Office: (0800) 371-371 Int'l: +64 4 528-6250 Mobile-NZ: 021 858-002 Australia: 03 8807-1406 IAXTel: 1700 528-6250 ========================
Hello, Has anybody experienced not sending audio from a PingTel xpressa phone? I can receive audio ok, but cannot send it. I have set mu-Law as the first codec. Alessandro
Jeremy Bogan
2004-Oct-20 16:31 UTC
[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 [ jeremy@segpub.com.au ] segment publishing - design.develop.host
James Sizemore
2004-Oct-21 09:48 UTC
[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. >