[asterisk-users] Unknown signalling method 'bri_cpe
Hello, Using libpri-1.4.7 and asterisk-1.6.0.1, I've got Unknown signalling method 'bri_cpe when module load chan_dahdi.so. Googling with chan_dahdi bri_net don't help much. Shall I upgrade to 1.6.1rcXXX to get 'bri_cpe support ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 Asterisks to one PBX - E1 conection
Hi Alejandro, thanks for reply, interesting and I'll try it. $300 isn't that much if it's reliable. Dubravko From: Alejandro Kauffmann [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 27, 2008 1:57:03 AM Subject: Re: [asterisk-users] 2 Asterisks to one PBX - E1 conection dubravko caric wrote: Hi all, I have a question regarding connection of two Asterisk servers to our PBX. Each Asterisk server has one PCI E1 card, and they are in failover mode with Linux HA. On our PBX we have only one E1 card towards Asterisk servers. My question is how to connect these two Asterisks to one E1 card on PBX, and when primary Asterisk server fails not to have to manually pull out E1 cable from primary server and plug it in secondary server in order to have active connection to E1 card on PBX. Is there some kind of splitter which, on one side can accept two E1 connections from Asterisks and on the other side one E1 link from PBX. This splitter must also recognize towards which one of two E1 links on Asterisk side it should send signals to. eg. when primary Asterisk fails this splitter should send signals to its eg. port 2 (connection towards secondary Asterisk). I would be most grateful if someone could provide me with a link to such products. Thanks Dubravko Don't know how well it works, but we've been looking at these: http://www.rhinoequipment.com/1portfail.html Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unknown signalling method 'bri_cpe
On Thu, Nov 27, 2008 at 09:24:31AM +0100, Olivier wrote: Hello, Using libpri-1.4.7 and asterisk-1.6.0.1, I've got Unknown signalling method 'bri_cpe when module load chan_dahdi.so. Googling with chan_dahdi bri_net don't help much. Shall I upgrade to 1.6.1rcXXX to get 'bri_cpe support ? Any chance asterisk is built without libpri support? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unknown signalling method 'bri_cpe
2008/11/27 Tzafrir Cohen [EMAIL PROTECTED] On Thu, Nov 27, 2008 at 09:24:31AM +0100, Olivier wrote: Hello, Using libpri-1.4.7 and asterisk-1.6.0.1, I've got Unknown signalling method 'bri_cpe when module load chan_dahdi.so. Googling with chan_dahdi bri_net don't help much. Shall I upgrade to 1.6.1rcXXX to get 'bri_cpe support ? Any chance asterisk is built without libpri support? How can I double check that ? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Any 1.6 SendFAX example ?
Hi, Do you have any example showing how to use SendFAX ? I can see several examples of ReceiveFAX but not a single one showing SendFAX. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] trunk peer not registering after migrating installation
I have an odd problem. I have just installed asterisk on an ubuntu box, and migrated the previous configuration of asterisk (on another ubuntu box) to this new server (scp -pr [EMAIL PROTECTED]:/etc/asterisk/* /etc/asterisk/) Asterisk worked fine on the old server, but on this server my SIP trunk peer does not login after initial server startup. sip show peers shows my phones registered OK, but the peer describing my SIP trunk does not even display: sip show peers Name/username HostDyn Nat ACL Port Status 204/204192.168.xxx.xxx D 2048 Unmonitored 203/203192.168.xxx.xxx D 2048 Unmonitored sip show registry sip.voipfone.co.uk:5060 45 Registered Thu, 27 Nov 2008 11:01:56:03 sip reload or restarting asterisk with /etc/init.d/asterisk restart fixes the problem and I get the following output: Name/username HostDyn Nat ACL Port Status 204/204192.168.xxx.xxx D 2048 Unmonitored 203/203192.168.xxx.xxx D 2048 Unmonitored voipfone/ 195.189.173.10 5060 OK (61 ms) sip show registry sip.voipfone.co.uk:5060 45 Registered Thu, 27 Nov 2008 11:05:28:02 sip.conf entry for the trunk [voipfone] type=friend secret=xx username= fromuser= fromdomain=sip.voipfone.co.uk host=sip.voipfone.co.uk insecure=very dtmfmode=rfc2833 context=fromvoipfone ;inbound calls falls in this context of dialplan disallow=all allow=ilbc ;allow=ulaw ;allow=alaw qualify=yes Any ideas warmly welcomed! Setting debug to level 9 isn't helping me out on this. John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mobile as FXO
I tried but no success. Do I have to add more to this?
Regards,
Irfan Malik
Manager MIS
TricastMedia
Cell +92 321-6099155
PH: +92 42 5785703-8 Ext: 196
Web: www.tcm.com.pk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas
Sent: Wednesday, November 26, 2008 7:43 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Mobile as FXO
I would try this:
exten = _.,1,Dial(Mobile/Nokia-7610/${EXTEN},60,KkTt) ; dials using mobile
nokia
7610
This should make the call Bridgeable/Transferrable.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Irfan Malik
Sent: Wednesday, November 26, 2008 8:38 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Mobile as FXO
These are lines from my extensions.conf
[phones]
; context for our phones
exten = 2001,1,Dial(SIP/2001)
exten = 2002,1,Dial(SIP/2002)
exten = 500,1,Answer()
exten = 500,2,Playback(demo-echotest)
exten = 500,3,Echo
exten = 500,4,Playback(demo-echodone)
exten = 500,5,Hangup
exten = _.,1,Dial(Mobile/Nokia-7610/${EXTEN}) ; dials using mobile nokia
7610
exten = _.,2,Hangup
Regards,
Irfan Malik
Manager MIS
TricastMedia
Cell +92 321-6099155
PH: +92 42 5785703-8 Ext: 196
Web: www.tcm.com.pk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas
Sent: Wednesday, November 26, 2008 7:32 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Mobile as FXO
What are the lines in your dialplan for using the Mobile line? For example
exten = NXX,1,Dial(Zap/g1/${EXTEN},60)
dials a local (7 digit) number using Zap Group 1, waiting 60 seconds for
connection.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Irfan Malik
Sent: Wednesday, November 26, 2008 8:25 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Mobile as FXO
How? Any hint?
Regards,
Irfan Malik
Manager MIS
TricastMedia
Cell +92 321-6099155
PH: +92 42 5785703-8 Ext: 196
Web: www.tcm.com.pk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, November 26, 2008 7:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Mobile as FXO
A little less whitespace please.
If I understand your question correctly, yes you can.
On Wed, Nov 26, 2008 at 9:10 AM, Irfan Malik [EMAIL PROTECTED] wrote:
Greetings List,
I have configured chan_mob for Nokia 7610. I can succefully dial from
softphone to mobile and land line numbers,
Softphone (PC) ===è Asterisk ==è FXO (Nokia 7610) ==è Destination Number
When call is established I have to use Nokia 7610 for conversation. Is it
possible to use softphone, dial via mobile phone and have conversation
using
softphone?
Regards,
Irfan Mali
Manager MIS
TricastMedia
Cell +92 321-6099155
PH: +92 42 5785703-8 Ext: 196
Web: www.tcm.com.pk
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--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] Unknown signalling method 'bri_cpe
output is: # strings /usr/lib/asterisk/modules/chan_dahdi.so | grep '^DAHDI Telephony' DAHDI Telephony DAHDI Telephony Driver ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any 1.6 SendFAX example ?
On Thu, Nov 27, 2008 at 1:03 PM, Olivier [EMAIL PROTECTED] wrote:
Hi,
Do you have any example showing how to use SendFAX ?
I can see several examples of ReceiveFAX but not a single one showing
SendFAX.
This is not from 1.6, but rather from callweaver attached to Asterisk 1.4.
When i'll finally switch to 1.6, i intend to just move those contexts
to Asterisk dialplan.
extensions.conf:
[fax_out]
exten = _X.,1,NoOp(--- sending fax to ${EXTEN} ---)
exten = _X.,n,SipDTMFMode(inband)
exten = _X.,n,TxFAX(${TIFF},caller,debug,ecm)
exten = _X.,n,Hangup()
exten = h,1,NoOp(--- done sending fax ---)
exten = h,n,NoOp(TX: REMOTESTATIONID is ${REMOTESTATIONID})
exten = h,n,NoOp(TX: FAXPAGES is ${FAXPAGES})
exten = h,n,NoOp(TX: FAXRESOLUTION is ${FAXRESOLUTION})
exten = h,n,NoOp(TX: FAXBITRATE is ${FAXBITRATE})
exten = h,n,NoOp(TX: PHASEESTATUS is ${PHASEESTATUS})
exten = h,n,NoOp(TX: PHASEESTRING is ${PHASEESTRING})
exten = h,n,NoOp(TX: DIALSTATUS is ${DIALSTATUS})
exten = h,n,System(${SCRIPT}/fax_out_end.php --status
${uniqueid_storage} --pages ${FAXPAGES} --resolution ${FAXRESOLUTION}
--bitrate ${FAXBITRATE} --phase
exten = failed,1,NoOp(--- failed sending fax ---)
Then, to send a fax, generate tiff file and call-file.
Snapshot of my PHP generating call-file from hylafax job:
$channel = 'SIP/'.$job['number'].'@asterisk-t38';
$destination =
array('context'='fax_out','extension'=$job['number'],'priority'='1');
$vars = array(
'LOCALSTATIONID' = 'CallWeaver-T38-TxFax',
'T38CALL'='1',
'TIFF'=$job['private']['tiff_file'],
);
$callerid = 'CallWeaver T38 TxFax';
$waittime = 180;
$deliver_time = NULL;
$filename = NULL;
$retries = array();
$callfile_dir = T38_CALLFILE_DIR.'/';
$result =
ast_originate_callfile($channel,$destination,$vars,$callerid,$waittime,$deliver_time,$filename,$retries,$callfile_dir);
Of course you'll need ast_originate_callfile which writes data to file
and then moves to correct dir. I would publish that, but it's full of
my constants and realted to much other libs..
Basically, you dial destination number (SIP/[EMAIL PROTECTED]) and send
local side of channel to fax_out,${NUMBER},1 which does SendFax.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Unknown signalling method 'bri_cpe
On Thu, Nov 27, 2008 at 12:58:53PM +0100, Olivier wrote: output is: # strings /usr/lib/asterisk/modules/chan_dahdi.so | grep '^DAHDI Telephony' DAHDI Telephony DAHDI Telephony Driver A snippet from channels/chan_dahdi.c: static const char tdesc[] = DAHDI Telephony Driver #ifdef HAVE_PRI w/PRI #endif #ifdef HAVE_SS7 w/SS7 #endif ; -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wellgate Asterisk
I got a Wellgate 3804A and need some hints:
Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate
Wellgate 3804A settings (Line1~Line4):
1. Sip Config
Mode: Proxy
Primary Proxy IP Address: *.131
Primary Proxy port: 5060
Line1 Number: 1002
2. Security Config
Line1 Account: 1002
Line1 Password: **
3. Line Configuration
Line1: Type=FXO, Hunting Group=2, Hot Line = 88621002
Asterisk settings:
users.conf:
[1002]
context = DID_1002
host = *.133
username = 1002
secret = **
trunkname = WellGate-1002 ; GUI metadata
hasiax = no
registeriax = no
hassip = yes
registersip = yes
trunkstyle = voip
hasexten = no
host = dynamic
disallow = all
allow = ulaw,alaw,gsm,g726,g729
extensions.conf
1002 = SIP/1002
...
[DID_1002]
exten = _88621002,1,NoOp(${CALLERID(num)})
exten = _88621002,n,Wait(1)
exten = _88621002,n,SayUnixTime
include = DID_1001_timeinterval_working day|${timeinterval_working day}
include = DID_1001_default
[DID_1001_default]
exten = s,1,NoOp,${CALLERID(num)}-${CALLERID(name)}
exten = s,n,Answer
exten = s,n,zapateller(nocallerid) ; torture telemarketers
exten = s,n,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = s,n,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten = s,n,Hangup
include = default
[DID_1001_timeinterval_working day]
exten = _6888,1,Goto(default|6888|1)
If I call in at line2, then I can hear the Time announcement and I can
dial during that announcement an extension number.
BTW, where can I find the additional sounds I had at an previous setup
(If you know the extension, ...), which should replace the SayUnixTime
I have no idea how to get dial out to work. Can anybody give me a hint,
please?
In Asterisk I see:
[Nov 27 20:58:00] NOTICE[5095]: chan_sip.c:9227 sip_reg_timeout:--
Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #102)
-- Got SIP response 486 Busy Here back from *.133
*CLI sip show peers
1002/1002 *.133D 5060 Unmonitored
*CLI sip show users
1002 **
DID_1002 No RFC3581
*CLI sip show registry
*.133:5060 1002 120 Request Sent
bye
Ronald
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Re: [asterisk-users] Mobile as FXO
On Thu, Nov 27, 2008 at 04:27:50PM +0500, Irfan Malik wrote: I tried but no success. Do I have to add more to this? What did you do? What did happen when you did that? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Disable Transfer
Hi All, I want to prevent transfer on based of user, means we can disable any user or peer to transfer calls in asterisk. Can any one helps how can we prevent transfer feature. I am using asterisk 1.4 branch. Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unknown signalling method 'bri_cpe
On Thu, Nov 27, 2008 at 11:04:50AM +0100, Olivier wrote: 2008/11/27 Tzafrir Cohen [EMAIL PROTECTED] On Thu, Nov 27, 2008 at 09:24:31AM +0100, Olivier wrote: Hello, Using libpri-1.4.7 and asterisk-1.6.0.1, I've got Unknown signalling method 'bri_cpe when module load chan_dahdi.so. Googling with chan_dahdi bri_net don't help much. Shall I upgrade to 1.6.1rcXXX to get 'bri_cpe support ? Any chance asterisk is built without libpri support? How can I double check that ? strings /usr/lib/asterisk/modules/chan_dahdi.so | grep '^DAHDI Telephony' -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wellgate Asterisk
Ronald Wiplinger (Lists) wrote: I got a Wellgate 3804A and need some hints: Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate Wellgate 3804A settings (Line1~Line4): Hi, as far as I can see, welltech also sells (embedded) asterix pbx units. Chances are, you might find a manual on the welltech website that describes how to integrate their flavour of Asterisk into the 3804. regards Eberhard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unknown signalling method 'bri_cpe
2008/11/27 Tzafrir Cohen [EMAIL PROTECTED] On Thu, Nov 27, 2008 at 12:58:53PM +0100, Olivier wrote: output is: # strings /usr/lib/asterisk/modules/chan_dahdi.so | grep '^DAHDI Telephony' DAHDI Telephony DAHDI Telephony Driver A snippet from channels/chan_dahdi.c: static const char tdesc[] = DAHDI Telephony Driver #ifdef HAVE_PRI w/PRI #endif So, I should have seen DAHDI Telephony Driver w/PRI, right ? I installed dahdi, lipbri and asterisk from source in this order, and obviously, I must have done something wrong. As chan_dahdi.so should include DAHDI Telephony Driver w/PRI, maybe I should have compiled libpri before dahdi or is ther something else ? #ifdef HAVE_SS7 w/SS7 #endif ; -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any 1.6 SendFAX example ?
Thanks for this detailed reply. I was trying to test SendFAX, ReceiveFAX as first on my way to Hylafax with either iaxmodem or t38modem. Have you tried any of those 2 (iaxmodem or t38modem) ? Which one would you pick ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wellgate Asterisk
El jue, 27-11-2008 a las 21:05 +0800, Ronald Wiplinger (Lists) escribió: I got a Wellgate 3804A and need some hints: Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate Wellgate 3804A settings (Line1~Line4): I've one wellgate 3804 (old version) with 4 fxo ports integrated with asterisk 1.4. Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esquina Edificio Barre #2 Primer Piso Telefono : +593 5 262 7815 Celular : +593 9 985 5138 International : +1 360 968 1701 e-mail: [EMAIL PROTECTED] www : http://www.telconet.net SIP : [EMAIL PROTECTED] Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any 1.6 SendFAX example ?
Olivier wrote: Thanks for this detailed reply. I was trying to test SendFAX, ReceiveFAX as first on my way to Hylafax with either iaxmodem or t38modem. Have you tried any of those 2 (iaxmodem or t38modem) ? Which one would you pick ? iaxmodem only does audio FAXing (for the present). t38modem only does T.38 FAXing. You pick the one you need. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints stopped working suddently
Just to follow-up, because this may one day be found by someone with the same issue, I fixed this: My problem was that my sip peers did not have a call-limit setup. For some (unknown to me) reason, hints only work for peers with a call-limit defined (if using realtime, that would mean something numerical, and not NULL). Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Wednesday, November 26, 2008 11:21 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Yes I did. Nothing changes, really. And it all looks good. What I don't get is why the status unavailable appears when the phone is disconnected, but the status inuse doesn't when on a call. That unavailable works fine is some sort of proof that everything is setup properly Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas Sent: Wednesday, November 26, 2008 11:18 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Have you tried doing core show hints and sip show peers before and after asterisk restart to see what if anything changes? _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Wednesday, November 26, 2008 10:11 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Not at all, I do everything with vi From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas Sent: Wednesday, November 26, 2008 8:51 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Hints stopped working suddently Do you use the Asterisk GUI? Changes from it can mess with contexts in the dialplan (extensions.conf) and the hints need to remain in the [internal] context. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Wednesday, November 26, 2008 6:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Hints stopped working suddently Hello, I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. I have hints setup (CLI show hints does show the hints, and they seem correct). But when I do dial using one of the SIP registrations, I don't see those hints being changed in the CLI (at verbose) like I used to. My hints keep on showing idle, even though I am making a call. Making this even weirder, if a phone falls off the grid I do get the subscription become unavailable. It's just the on call hint that does not seem to work. So it seems not to be a firewall/routing issue. I don't think it's the phones, since Asterisk doesn't seem to update it's internal hint (show hints command) when I dial out or get a call. Using Asterisk 1.4.22 and Polycom 3.1.1 firmware (latest). And I restarted asterisk just in case, no help. Regards, Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] originate problem
Hi there! Trying to originate and dial a number using Zap-8, used to work, but now it just fails. I enabled all debug I found in the source-code and this is the output from asterisk. Can someone understand something from the debug-output what is wrong and direct me to what the problem might be? The setup is correct, trust me, it worked some hours ago, haven't changed anything. Just dialing again and again to test... sometimes the Zap-8 line does not hangup. But I thought restarting asterisk would hang it up? Maybe it's still off hook. ? Thanks, Johan [Nov 27 16:46:25] DEBUG[907] manager.c: Manager received command 'Originate' [Nov 27 16:46:25] DEBUG[907] chan_zap.c: Using channel 8 [Nov 27 16:46:25] DEBUG[907] chan_zap.c: Dialing '0734414119' [Nov 27 16:46:25] DEBUG[907] chan_zap.c: Deferring dialing... [Nov 27 16:46:25] DEBUG[907] devicestate.c: Notification of state change to be queued on device/channel Zap/8-1 [Nov 27 16:46:25] DEBUG[907] devicestate.c: Notification of state change to be queued on device/channel Zap/8 [Nov 27 16:46:25] DEBUG[877] devicestate.c: No provider found, checking channel drivers for Zap - 8-1 [Nov 27 16:46:25] DEBUG[877] devicestate.c: Changing state for Zap/8-1 - state 0 (Unknown) [Nov 27 16:46:25] DEBUG[877] devicestate.c: No provider found, checking channel drivers for Zap - 8 [Nov 27 16:46:25] DEBUG[877] devicestate.c: Changing state for Zap/8 - state 2 (In use) [Nov 27 16:46:25] DEBUG[902] app_queue.c: Device 'Zap/8-1' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. [Nov 27 16:46:25] DEBUG[902] app_queue.c: Device 'Zap/8' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 27 16:46:26] DEBUG[907] chan_zap.c: Exception on 27, channel 8- this doesn't look good... what does it mean? :-O [Nov 27 16:46:26] DEBUG[907] chan_zap.c: option_debug=100 [Nov 27 16:46:26] DEBUG[907] chan_zap.c: Got event Hook Transition Complete(12) on channel 8 (index 0) [Nov 27 16:46:26] DEBUG[907] chan_zap.c: Sent deferred digit string: T0734414119w [Nov 27 16:46:30] DEBUG[907] chan_zap.c: Exception on 27, channel 8 [Nov 27 16:46:30] DEBUG[907] chan_zap.c: option_debug=100 [Nov 27 16:46:30] DEBUG[907] chan_zap.c: Got event Dial Complete(9) on channel 8 (index 0) [Nov 27 16:46:30] DEBUG[907] chan_zap.c: Enabled echo cancellation on channel 8 The call obivously failed... very strange. Even if I restart asterisk it is still not working... :( [Nov 27 16:46:56] DEBUG[907] channel.c: Hanging up channel 'Zap/8-1' [Nov 27 16:46:56] DEBUG[907] chan_zap.c: zt_hangup(Zap/8-1) [Nov 27 16:46:56] DEBUG[907] chan_zap.c: Hangup: channel: 8 index = 0, normal = 27, callwait = -1, thirdcall = -1 [Nov 27 16:46:56] DEBUG[907] chan_zap.c: disabled echo cancellation on channel 8 [Nov 27 16:46:56] DEBUG[907] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/8-1 [Nov 27 16:46:56] DEBUG[907] chan_zap.c: Updated conferencing on 8, with 0 conference users [Nov 27 16:46:56] VERBOSE[907] logger.c: -- Hungup 'Zap/8-1' ___ Johan Sandgren Svep Design Center AB Phone +46 46 192 722 Mobile +46 70 173 4152 Box 1233, 221 05 Lund, Sweden E-mail [EMAIL PROTECTED] Website www.svep.sehttp://www.svep.se/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] originate problem
On Thu, Nov 27, 2008 at 05:02:17PM +0100, Johan Sandgren wrote: Hi there! Trying to originate and dial a number using Zap-8, used to work, but now it just fails. I enabled all debug I found in the source-code and this is the output from asterisk. Can someone understand something from the debug-output what is wrong and direct me to what the problem might be? The setup is correct, trust me, it worked some hours ago, haven't changed anything. Just dialing again and again to test... sometimes the Zap-8 line does not hangup. But I thought restarting asterisk would hang it up? Maybe it's still off hook. ? What device? What version of Asterisk? Thanks, Johan [Nov 27 16:46:25] DEBUG[907] manager.c: Manager received command 'Originate' [Nov 27 16:46:25] DEBUG[907] chan_zap.c: Using channel 8 [Nov 27 16:46:25] DEBUG[907] chan_zap.c: Dialing '0734414119' [Nov 27 16:46:25] DEBUG[907] chan_zap.c: Deferring dialing... [Nov 27 16:46:25] DEBUG[907] devicestate.c: Notification of state change to be queued on device/channel Zap/8-1 [Nov 27 16:46:25] DEBUG[907] devicestate.c: Notification of state change to be queued on device/channel Zap/8 [Nov 27 16:46:25] DEBUG[877] devicestate.c: No provider found, checking channel drivers for Zap - 8-1 [Nov 27 16:46:25] DEBUG[877] devicestate.c: Changing state for Zap/8-1 - state 0 (Unknown) [Nov 27 16:46:25] DEBUG[877] devicestate.c: No provider found, checking channel drivers for Zap - 8 [Nov 27 16:46:25] DEBUG[877] devicestate.c: Changing state for Zap/8 - state 2 (In use) [Nov 27 16:46:25] DEBUG[902] app_queue.c: Device 'Zap/8-1' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. [Nov 27 16:46:25] DEBUG[902] app_queue.c: Device 'Zap/8' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 27 16:46:26] DEBUG[907] chan_zap.c: Exception on 27, channel 8- this doesn't look good... what does it mean? :-O [Nov 27 16:46:26] DEBUG[907] chan_zap.c: option_debug=100 [Nov 27 16:46:26] DEBUG[907] chan_zap.c: Got event Hook Transition Complete(12) on channel 8 (index 0) [Nov 27 16:46:26] DEBUG[907] chan_zap.c: Sent deferred digit string: T0734414119w [Nov 27 16:46:30] DEBUG[907] chan_zap.c: Exception on 27, channel 8 [Nov 27 16:46:30] DEBUG[907] chan_zap.c: option_debug=100 [Nov 27 16:46:30] DEBUG[907] chan_zap.c: Got event Dial Complete(9) on channel 8 (index 0) [Nov 27 16:46:30] DEBUG[907] chan_zap.c: Enabled echo cancellation on channel 8 The call obivously failed... very strange. Even if I restart asterisk it is still not working... :( [Nov 27 16:46:56] DEBUG[907] channel.c: Hanging up channel 'Zap/8-1' [Nov 27 16:46:56] DEBUG[907] chan_zap.c: zt_hangup(Zap/8-1) [Nov 27 16:46:56] DEBUG[907] chan_zap.c: Hangup: channel: 8 index = 0, normal = 27, callwait = -1, thirdcall = -1 [Nov 27 16:46:56] DEBUG[907] chan_zap.c: disabled echo cancellation on channel 8 [Nov 27 16:46:56] DEBUG[907] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/8-1 [Nov 27 16:46:56] DEBUG[907] chan_zap.c: Updated conferencing on 8, with 0 conference users [Nov 27 16:46:56] VERBOSE[907] logger.c: -- Hungup 'Zap/8-1' ___ Johan Sandgren Svep Design Center AB Phone +46 46 192 722 Mobile +46 70 173 4152 Box 1233, 221 05 Lund, Sweden E-mail [EMAIL PROTECTED] Website www.svep.sehttp://www.svep.se/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: cdr_addon_mysql.so did not register itselfduringload
On Thu, 28 Dec 2006 12:34:46 -0600, Savoy, Kevin - Williston, ND wrote: checking for mysql_init in -lmysqlclient... no What do I need to make that say yes? You need to read config.log and check _why_ the link fails. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints stopped working suddently
Mike, I don't want to be a smart ass, but (as you claimed) if you didn't change anything I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. how was it working before ? I really want to know, as there may be something else going on in the background. Julian. Mike wrote: Just to follow-up, because this may one day be found by someone with the same issue, I fixed this: My problem was that my sip peers did not have a call-limit setup. For some (unknown to me) reason, hints only work for peers with a call-limit defined (if using realtime, that would mean something numerical, and not NULL). Mike *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Mike *Sent:* Wednesday, November 26, 2008 11:21 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Yes I did. Nothing changes, really. And it all looks good. What I don't get is why the status unavailable appears when the phone is disconnected, but the status inuse doesn't when on a call. That unavailable works fine is some sort of proof that everything is setup properly… Mike *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, November 26, 2008 11:18 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Have you tried doing “core show hints” and “sip show peers” before and after asterisk restart to see what if anything changes? *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Mike *Sent:* Wednesday, November 26, 2008 10:11 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Not at all, I do everything with vi *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, November 26, 2008 8:51 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Hints stopped working suddently Do you use the Asterisk GUI? Changes from it can mess with contexts in the dialplan (extensions.conf) and the hints need to remain in the [internal] context. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Mike *Sent:* Wednesday, November 26, 2008 6:33 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* [asterisk-users] Hints stopped working suddently Hello, I've had Asterisk and Polycom phones work perfectly with hints for the last 6 months. Suddently, I realize they've stopped working in the last few days. I haven't changed the configuration in any way. I have hints setup (CLI show hints does show the hints, and they seem correct). But when I do dial using one of the SIP registrations, I don't see those hints being changed in the CLI (at verbose) like I used to. My hints keep on showing idle, even though I am making a call. Making this even weirder, if a phone falls off the grid I do get the subscription become unavailable. It's just the on call hint that does not seem to work. So it seems not to be a firewall/routing issue. I don't think it's the phones, since Asterisk doesn't seem to update it's internal hint (show hints command) when I dial out or get a call. Using Asterisk 1.4.22 and Polycom 3.1.1 firmware (latest). And I restarted asterisk just in case, no help. Regards, * * * * *Mike* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pick up IAX2 calls
You can either add that feature to chan_iax2.c or pay someone to add that feature to chan_iax2.c. Bruno Castelo Branco wrote: Somebody know some work around for it? I still trying to find a solution but nothing seems to work thanks Eric ManxPower Wieling wrote: The problem is that IAX2 does not seem to support call pickup. Bruno Castelo Branco wrote: hi I'm using only IAX extensions and inserted callgroup=1 and callpickup=1 for all IAX extensions in iax.conf. Didn't works for while. thanks Tim Panton wrote: I think it doesn't work across channel types. So it works (if I recall correctly) in IAX or in SIP or in ZAP, but not in mixture. I think that if you have a Dial() that rings several extens, then any of the technologies involved can pickup with *8 So if you have Dial(IAX/fredSIP/billzap/mark) then someone in the same group as fred can pickup with IAX and someone in the same group as bill can pickup with SIP etc. So it's an asterisk thing, not an IAX thing per-se. Tim. P.S. (you could try putting in a dummy 'fred' entry into Dial and iax.conf.) T. On 25 Nov 2008, at 01:09, Bruno Castelo Branco wrote: hi thanks Luis , but doesn't work. For SIP extensions works well *8, but for IAX a tried *8 and ** + iax extension and didn't works Luis Morales wrote: Try with ** + iax extension Regards, Luis Morales On Mon, Nov 24, 2008 at 10:20 PM, Bruno Castelo Branco [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Somebody knows if pickup call works with IAX2? I enable *8 in features.conf, but doesn't works with IAX2 extensions. Any idea? thanks -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any 1.6 SendFAX example ?
On Thu, Nov 27, 2008 at 4:39 PM, Olivier [EMAIL PROTECTED] wrote: Thanks for this detailed reply. I was trying to test SendFAX, ReceiveFAX as first on my way to Hylafax with either iaxmodem or t38modem. Have you tried any of those 2 (iaxmodem or t38modem) ? Which one would you pick ? We had IAXmodem with Hylafax installation base, and we sent our faxes out trough PRI. Then we switched to pure SIP, but were unable to get T38modem to work with our provider. So, we wrote a wrapper for Hylafax that grabs processed tiff file from outgoing spool and generates call file for Callweaver (which sends trough Asterisk with T38 passtrough). So, if you have PRI ir analogue lines, use IAXmodem, otherwise you have to do either T38modem or SendFax. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MOH Realtime
Is working on 1.6.0.1?? someone was able to make it work? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] force channel hangup
Hi guys, I have 1 zap channel in my house shared among couple people. If someone dials 911, I want that zap channel to be disconnected right away to make way for the 911 call. I dug through voip-info.org and didn't find much. Any hints? kel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Windows Mobile 6 SIP client: Remote host can't match request NOTIFY to call
I'm trying to get my Windows Mobile 6 phone working as an asterisk client. Overall things are working well. However, I regularly get the following message: [Nov 27 21:57:28] WARNING[4507]: chan_sip.c:12892 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. From what I've read, the client doesn't subscribe to MWI but gets a notify event - which it rejects. The voicemail notifications ARE working on the device. Any way to get rid of this message (while keeping the MWI on the phone)? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities
I did a test yesterday and did 1,000 registrations to Asterisk using SIPP. I did the register test since I am using the realtime DB and asterisk does periodic quesries to it for each registered user. Although Asterisk continued to function as usuall, it was in a steady loop querying the DB for the 1,000 users. OK, you convinced me to look at some front end to it. There are mainly three front ends mentioed here: OpenSer, SipExpress and FreeSwitch. Is there some comparison available which will save me from testing all three of them? Is there one which is more used than the others? (so it has more public QA :-) Thanks! __Yehavi: 2008/11/24 Steve Totaro [EMAIL PROTECTED] Fronting with OpenSER or FS, you should have no problems providing you plan to use SIP extensions. What is critical are the max simultaneous trunks you are going to use. I would go TDM although universities have good bandwidth, and SUPERIOR bandwidth between others. I would think a TDM DS3 or two just to be safe. It should be pretty trivial besides gotchas, like cat3 to the rooms, although channel banks may be an even better solution if phones are already in place. Then you just use SIP when needed or wanted, and Asterisk is simple, although more costly. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Fri, Nov 21, 2008 at 6:24 PM, Wilton Helm [EMAIL PROTECTED] wrote: Yet another option is a commercial system with in-house staff. I used to maintain a NEC (NEAX 2400) for many years. I went to factory training and had total responsibility for it. Some manufacturers discourage or prevent this, but others are open to it. There are also 3rd party organizations (such as Source) that can supply parts and even expertise for those going that direction. Whether the result would be higher availability than Asterisk, I don't know. Given I'm both a telco guy and a computer guru (CS degree) I'd probably go the Asterisk route myself, because its open and I would have more control. Wilton and bug fixes than any commercial product sold in the intra-industrial channel ... and they won't charge you a $30,000 license fee for the upgrade. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH Realtime
Hi Sebastian, http://bugs.digium.com/view.php?id=11196 Nguyễn Đình Trung --- QiS Technologies, ltd. Tel: 0168 528 7522 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Solved] Wellgate Asterisk
Guillermo Salas M. wrote: El jue, 27-11-2008 a las 21:05 +0800, Ronald Wiplinger (Lists) escribió: I got a Wellgate 3804A and need some hints: Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate Wellgate 3804A settings (Line1~Line4): I've one wellgate 3804 (old version) with 4 fxo ports integrated with asterisk 1.4. Regards, I could solve it! I had to add routing in the 3804A. Now both, dialin and dialout is working. bye Ronald ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Windows Mobile 6 SIP client: Remote host can't match request NOTIFY to call
Have you snniffed the packages? It seems to be some kind of difrerence on the notify, try to sniff a packet ok and then one with error Enviado desde mi iPhone El 28/11/2008, a las 01:01 a.m., OCG Technical Support [EMAIL PROTECTED] escribió: I’m trying to get my Windows Mobile 6 phone working as an asterisk c lient. Overall things are working well. However, I regularly get th e following message: [Nov 27 21:57:28] WARNING[4507]: chan_sip.c:12892 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED] '. Giving up. From what I’ve read, the client doesn’t subscribe to MWI but gets a notify event – which it rejects. The voicemail notifications ARE working on the device. Any way to get rid of this message (while keeping the MWI on the phone)? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH Realtime
Thanks for the answer i did everything that is on that issue but is not working, do you have it working? Every thing else im doing real time but moh never check the db if i try the command realtime load ... I get the values just fine, but they are never realy load to the memory classes Sorry the spelling but im writing on the phone with spanish corrector :) Enviado desde mi iPhone El 28/11/2008, a las 02:29 a.m., dinhtrung [EMAIL PROTECTED] escribió: Hi Sebastian, http://bugs.digium.com/view.php?id=11196 Nguyễn Đình Trung --- QiS Technologies, ltd. Tel: 0168 528 7522 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anonymous callerid
Hi All I have one issue regarding override callerid when i have anonymous call. I have added PAI in sip header and also set sendrpid = yes in sip.conf but the callerid is not overriding while i am sending call to three digit calling like 911. please give some idea and help for this issue! I am using asterisk 1.4 branch. thanks in advance!! Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
