Re: [asterisk-users] CallerID in UK on TalkTalk - different to BT?
Its strange, if I reboot my Asterisk you get no callerid. But then if you do a reload of the config then callerid comes back. any ideas why this could happen? Angus - Original Message - From: "Steve Kennedy" <[EMAIL PROTECTED]> To: Sent: Saturday, July 08, 2006 12:37 PM Subject: Re: [asterisk-users] CallerID in UK on TalkTalk - different to BT? > On Sat, Jul 08, 2006 at 10:59:49AM +0100, Angus Comber wrote: > > > I had an Asterisk installation working fine for CallerID on BT analog lines > > using a Digium analog 4 port card. However, user switched to TalkTalk > > without telling me and CallerID no longer works. However, if you connect a > > UK CallerID capable phone into one of these analog lines directly you do see > > the CallerID. > > Does anyone know how to tweak the settings for Talk Talk. Talk Talk have > > basically taken over the line rental - and they supply everything including > > the CLIP (CallerID) service now. > > Just to be clear CallerID was working fine before when line rental supplied > > by BT. > > Even though the line has been taken over by TalkTalk, it's still a BT > line off a BT Exchange so the initial leg of the call (or final > depending on your point of view) is still BT. > > Steve > > -- > NetTek Ltd UK mob +44-(0)7775 755503 > UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 > Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] > Euro Tech News Blog http://eurotechnews.blogspot.com > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallerID in UK on TalkTalk - different to BT?
Hello I had an Asterisk installation working fine for CallerID on BT analog lines using a Digium analog 4 port card. However, user switched to TalkTalk without telling me and CallerID no longer works. However, if you connect a UK CallerID capable phone into one of these analog lines directly you do see the CallerID. Does anyone know how to tweak the settings for Talk Talk. Talk Talk have basically taken over the line rental - and they supply everything including the CLIP (CallerID) service now. Just to be clear CallerID was working fine before when line rental supplied by BT. Some of zapata.conf: usecallerid=yes cidsignalling=v23 ; added for UK CLI detection cidstart=polarity ; added for UK CLI detection Angus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How would you go about calling a list of numbers and 'speaking' a message?
Hello I have been asked by a client to process a list of telephone numbers. Asterisk should call each number in turn and if the recipient of the call answers, play a message - eg from a wav. How would I go about doing that? Angus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What do I need to setup Asterisk with an H323 client?
Hello I want to test asterisk with an H323 client. In Windows XP there is phone dialer which can use H323. In Phone dialer I set H323 Line for phone calls and Internet calls. In Phone and Modem properties H323 provider I set: H.323 gatekeeper: 192.168.0.20 (asterisk on my LAN) Log on using my phone number - 400 Gatekeeper registration state says 'Not Registered' On the asterisk I have a h323.conf file like this: [general] port=1720 bindaddr=0.0.0.0 disallow=all allow=ulaw dtmfmode=rfc2833 context=default [400] type=friend context=default callerid="400" <400> I have in my extensions.conf: exten => 400,1,Dial(H323/400) exten => 400,2,Hangup But you guessed it, it doesn't work! I haven't installed anything extra on my asterisk box. Do I need to install something? What? I have seen mention of oh323 and oh323.conf and h323.conf? What do I need? Or what should I use? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why can't I dial - just using SIP internally
Hello
I have setup a couple of sip accounts - here is my sip.conf:
context=default
disallow=all
allow=ulaw
allow=alaw
allow=gsm
[200]
username=200
type=friend
secret=1234
port=5060
nat=never
[EMAIL PROTECTED]
dtmfmode=rfc2833
context=default
callerid="Angus" <200>
host=dynamic
insecure=very
group=1
callgroup=1
pickupgroup=1
[201]
username=201
type=friend
secret=1234
port=5060
nat=never
dtmfmode=rfc2833
context=default
callerid="Lisa" <201>
host=dynamic
insecure=very
group=1
callgroup=1
pickupgroup=1
my extensions.conf:
[frompstnanalog]
exten => 787367,1,Dial(SIP/200,1)
exten => 787367,2,Voicemail(su200)
exten => 787367,3,Hangup
[default]
;exten => _X.,1,Dial(ZAP/g1/${EXTEN},20,Ttm)
;exten => _X.,2,Hangup
exten => _2XX,1,Dial(SIP/${EXTEN},20,Ttm)
exten => _2XX,2,Voicemail(su${EXTEN})
exten => _2XX,3,Hangup
exten => *97,1,Answer
exten => *97,2,VoicemailMain([EMAIL PROTECTED])
exten => *97,3,Hangup
I have setup two IP phones, they register OK but cannot dial each other. I
had to switch on sip debug to get anything on the asterisk console:
pbx*CLI>
<-- SIP read from 192.168.0.21:5060:
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK-17vz3vxf4uhz;rport
From: "Angus" ;tag=oa5ljlnorj
To:
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Max-Forwards: 70
Contact:
P-Key-Flags: keys="3"
User-Agent: snom190-3.56m
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 342
v=0
o=root 2065976712 2065976712 IN IP4 192.168.0.21
s=call
c=IN IP4 192.168.0.21
t=0 0
m=audio 1 RTP/AVP 0 8 3 18 4 9 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:9 g722/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (18 headers 16 lines)---
Using INVITE request as basis request -
[EMAIL PROTECTED]
Sending to 192.168.0.21 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.0.21:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.0.21:5060;branch=z9hG4bK-17vz3vxf4uhz;rport;received=192.168.0.21
From: "Angus" ;tag=oa5ljlnorj
To: ;tag=as7203b20e
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="24b5d1a5"
Content-Length: 0
---
Scheduling destruction of call '[EMAIL PROTECTED]' in
15000 ms
Found user '200'
pbx*CLI>
<-- SIP read from 192.168.0.21:5060:
ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK-17vz3vxf4uhz;rport
From: "Angus" ;tag=oa5ljlnorj
To: ;tag=as7203b20e
Call-ID: [EMAIL PROTECTED]
CSeq: 1 ACK
Max-Forwards: 70
Contact:
Content-Length: 0
--- (9 headers 0 lines)---
pbx*CLI>
<-- SIP read from 192.168.0.21:5060:
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK-ass0mb36ivsw;rport
From: "Angus" ;tag=oa5ljlnorj
To:
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
Max-Forwards: 70
Contact:
P-Key-Flags: keys="3"
User-Agent: snom190-3.56m
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Proxy-Authorization: Digest
username="200",realm="asterisk",nonce="24b5d1a5",uri="sip:[EMAIL
PROTECTED];user=phone",response="a5598b627eb4c3bad2084bd553daad3f",algorithm=md5
Content-Type: application/sdp
Content-Length: 342
v=0
o=root 2065976712 2065976712 IN IP4 192.168.0.21
s=call
c=IN IP4 192.168.0.21
t=0 0
m=audio 1 RTP/AVP 0 8 3 18 4 9 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:9 g722/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (19 headers 16 lines)---
Using INVITE request as basis request -
[EMAIL PROTECTED]
Sending to 192.168.0.21 : 5060 (non-NAT)
Found user '200'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 9
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.21:1
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format g729
Found description format g723
Found description format g722
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10f
(g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xe
(gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 201 in default (domain 192.168.0.20)
Reliably Tra
Re: [Asterisk-Users] *8 and group pickup not working
When I added group=1 callgroup=1 pickupgroup=1 under each extension then it worked. I assume it is the pickupgroup=1 that did it. I will experiment to see. Angus - Original Message - From: "Alberto Risco" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" ; <[EMAIL PROTECTED]> Sent: Monday, October 10, 2005 7:14 PM Subject: RE: [Asterisk-Users] *8 and group pickup not working I don't know if this will help you, but we had the same problem, we also have Polycom 500s and I changed the pickupexten to *9 (anything other than *8), because I read somewhere that for some reason Asterisk has a problem with this feature and *8. It worked for us. Alberto -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angus Comber Sent: Sunday, October 09, 2005 2:48 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] *8 and group pickup not working No that's not problem. On my current configs I get: Oct 9 20:43:18 NOTICE[2990]: chan_sip.c:7455 handle_request: Nothing to pick up every time I try *8 Why does the phone think there is nothing to pickup? Angus - Original Message - From: "Alan Harrison" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Sunday, October 09, 2005 2:35 PM Subject: Re: [Asterisk-Users] *8 and group pickup not working On Sun, 9 Oct 2005 21:32, Angus Comber wrote: Hi I have Polycom 600s and 500s but I find that we need to dial *8 then send. If we pickup then dial *8 the phone or Asterisk re-aranges it to 8*. Likewise with *97 and *98 foes to 9*7 and 9*8. This might help. Hello I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom IP300 phones. My config files look like this: features.conf pickupextn = *8 zapata.conf context=frompstnisdn group=1 callgroup=1 pickupgroup=1 I also edited sip.conf like this: group=1 callgroup=1 pickupgroup=1 But on internal and incoming calls if I dial *8 from any phone I cannot pickup. Do I need to add anything to extensions.conf? do something else. I also tested with a Snom 190 and that cannot pickup using *8 either! Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Alan Harrison PABX Advisory Services Pty Ltd PH 02 9893 7888 Email [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *8 and group pickup not working
No that's not problem. On my current configs I get: Oct 9 20:43:18 NOTICE[2990]: chan_sip.c:7455 handle_request: Nothing to pick up every time I try *8 Why does the phone think there is nothing to pickup? Angus - Original Message - From: "Alan Harrison" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Sunday, October 09, 2005 2:35 PM Subject: Re: [Asterisk-Users] *8 and group pickup not working On Sun, 9 Oct 2005 21:32, Angus Comber wrote: Hi I have Polycom 600s and 500s but I find that we need to dial *8 then send. If we pickup then dial *8 the phone or Asterisk re-aranges it to 8*. Likewise with *97 and *98 foes to 9*7 and 9*8. This might help. Hello I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom IP300 phones. My config files look like this: features.conf pickupextn = *8 zapata.conf context=frompstnisdn group=1 callgroup=1 pickupgroup=1 I also edited sip.conf like this: group=1 callgroup=1 pickupgroup=1 But on internal and incoming calls if I dial *8 from any phone I cannot pickup. Do I need to add anything to extensions.conf? do something else. I also tested with a Snom 190 and that cannot pickup using *8 either! Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Alan Harrison PABX Advisory Services Pty Ltd PH 02 9893 7888 Email [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *8 and group pickup not working
I tried both
exten => *8,1,PickUP()
and
exten => *8,1,PickUp(1)
But got:
-- Accepting voice call from '7768385144' to '787367' on channel 0/1,
span 1
-- Executing Dial("Zap/1-1", "SIP/200&SIP/202|20") in new stack
-- Called 200
-- Called 202
-- SIP/202-f041 is ringing
-- SIP/200-0f37 is ringing
Oct 9 20:03:53 NOTICE[2990]: chan_sip.c:7455 handle_request: Nothing to
pick up
-- Channel 0/1, span 1 got hangup
== Spawn extension (frompstnisdn, 787367, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Executing Dial("SIP/201-f7db", "SIP/202|20|Ttm") in new stack
-- Called 202
-- Started music on hold, class 'default', on SIP/201-f7db
-- SIP/202-21a3 is ringing
Oct 9 20:04:09 NOTICE[2990]: chan_sip.c:7455 handle_request: Nothing to
pick up
-- Stopped music on hold on SIP/201-f7db
== Spawn extension (default, 202, 1) exited non-zero on 'SIP/201-f7db'
Any ideas?
Angus
- Original Message -
From: "Tzafrir Cohen" <[EMAIL PROTECTED]>
To:
Sent: Sunday, October 09, 2005 1:31 PM
Subject: Re: [Asterisk-Users] *8 and group pickup not working
On Sun, Oct 09, 2005 at 12:32:12PM +0100, Angus Comber wrote:
Hello
I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom
IP300 phones.
I figure you use bristuff. Are you aware of app_pickup that comes with
it?
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il | | a Mutt's
[EMAIL PROTECTED] | | best
ICQ# 16849755 | | friend
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Re: [Asterisk-Users] *8 and group pickup not working
Yes sadly a typo on my part. It is pickupexten in features.conf Any other ideas? Angus - Original Message - From: "Guido Hecken" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Sunday, October 09, 2005 12:54 PM Subject: RE: [Asterisk-Users] *8 and group pickup not working Shouldn't it be pickupexten = *8 instead of pickupextn = *8 ? Regards Guido Hecken Hello I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom IP300 phones. My config files look like this: features.conf pickupextn = *8 zapata.conf context=frompstnisdn group=1 callgroup=1 pickupgroup=1 I also edited sip.conf like this: group=1 callgroup=1 pickupgroup=1 But on internal and incoming calls if I dial *8 from any phone I cannot pickup. Do I need to add anything to extensions.conf? do something else. I also tested with a Snom 190 and that cannot pickup using *8 either! Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] *8 and group pickup not working
Hello I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom IP300 phones. My config files look like this: features.conf pickupextn = *8 zapata.conf context=frompstnisdn group=1 callgroup=1 pickupgroup=1 I also edited sip.conf like this: group=1 callgroup=1 pickupgroup=1 But on internal and incoming calls if I dial *8 from any phone I cannot pickup. Do I need to add anything to extensions.conf? do something else. I also tested with a Snom 190 and that cannot pickup using *8 either! Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiFi Phones
wireless generally struggles with brick walls. - Original Message - From: "Matt Love" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Friday, October 07, 2005 9:55 AM Subject: RE: [Asterisk-Users] WiFi Phones Hi I have a Zyxel 2000W wifi phone, setup is easy and quick to perform. However I have found the range less that satisfactory. I have a Cisco 1200 AP and our wireless laptop devices can acccess the network fine, however the Zyxel is pretty rubbish. For example I can be 5 metres away with only a single brick wall in the way and hardly have signal. It could be this particular handset has a problem. I would be interested to see if anyone else has a similar experience or could it be my phone? Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ADEGOKE ARUNA Sent: 07 October 2005 09:41 To: 'Andy Hamilton'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] WiFi Phones Can you try zyxel. I has graphical interface to do the configuration. goksie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Hamilton Sent: Friday, October 07, 2005 4:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi Phones Anyone have good words to say about any of the WiFi handsets currently available? The UTStarCom F1000 (an 802.11b device) works pretty well. It's about half the $$$ of a Cisco 7920 (which are also pretty nice), but it seems like most of the config is done from the keypad. There is a TFTP option, but it seems that isn't quite perfect. You could check the manual (I programmed the unit without that, except to find that the default password is 88). The unit, I'm guessing, was designed somewhere in Asia, and the language translation shows it a little bit. Sound quality seems pretty good for the few calls I've passed through it. I only have one AP in my house, so I can't comment on roaming. The headset for my cell phone is stereo, and I think the phone would be most happy with a standard 3 conductor plug, but I imagine a headset on a phone is a headset on a phone. The keypad is a touch small, and sometimes I hit the wrong key (and my fingers aren't terribly fat). I also seemed to have a problem transferring calls (using the built in transfer function -- # should still work). Despite many vendors' pages saying that it does 802.1x authentication, it sure looks like WEP is the only available "security" option. Overall: I would recommend purchasing one, for testing at the very least. They are well priced and of good quality. Battery life seems to be pretty good, too. -A ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk/Debian/VIA EPIA M Howto
Your link doesn't seem to work. Angus - Original Message - From: "Cameron Steadman" <[EMAIL PROTECTED]> To: Sent: Thursday, October 06, 2005 4:17 PM Subject: [Asterisk-Users] Asterisk/Debian/VIA EPIA M Howto I have written a step-by-step setup for installing Asterisk on Debian using the VIA EPIA M platform. It is oriented to the Linux novice (myself being one). Feel free to use it :) http://www.steady-com.com/asterisk/debian-install.html ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel TDM questions
Could you not just ignore the first answer and watch out for the answer when the remote end picks up? Angus - Original Message - From: Chee Foong To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, October 05, 2005 11:35 AM Subject: RE: [Asterisk-Users] Zaptel TDM questions Yes, we have an applications that needs to detect the actual answer of the call not when it is ringing. CCF -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Angus ComberSent: Friday, September 30, 2005 19:18To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Zaptel TDM questions I think the Asterisk must answer the call to be able to then dial out on the second port. This is what happens on any other PBX I have worked with in this sort of scenario. Is this a problem for you? Angus - Original Message - From: Chee Foong To: asterisk-users@lists.digium.com Sent: Friday, September 30, 2005 10:20 AM Subject: [Asterisk-Users] Zaptel TDM questions Hello, I have a TDM04B. I make a call into the first port of the card. Once asterisk receive the call, it will make another call out using the second port. From what i have observerd as soon as the called party on the second port starts ringing asterisk show the following : -- Zap/2-1 answered Zap/1-1 Any idea why asterisk thinks the call has been answered while actually the phone is still ringing? Anybody know how to avoid asterisk to answer the call while ringing? Also, I have no Answer or any Playback command in the dial plan before making a call out of second port. I have also try setting callprogress to yes/no but the results are the same. Thanks CCF ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] agi-test.agi question - wierd results
Hello
I am starting to learn AGI. I have setup an extension to play the
agi-test.agi perl script and the output I get is this on console:
On Polycom 300:
-- Executing Answer("SIP/200-72d2", "") in new stack
-- Executing AGI("SIP/200-72d2", "agi-test.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-test.agi
-- Playing 'digits/1' (language 'en')
-- Playing 'digits/hundred' (language 'en')
-- Playing 'digits/90' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/million' (language 'en')
-- Playing 'digits/8' (language 'en')
-- Playing 'digits/hundred' (language 'en')
-- Playing 'digits/30' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/thousand' (language 'en')
-- Playing 'digits/4' (language 'en')
-- Playing 'digits/hundred' (language 'en')
-- Playing 'digits/60' (language 'en')
-- Playing 'digits/5' (language 'en')
On other handsets:
-- Executing Answer("SIP/201-4415", "") in new stack
-- Executing AGI("SIP/201-4415", "agi-test.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-test.agi
-- Playing 'digits/1' (language 'en')
-- Playing 'digits/hundred' (language 'en')
-- Playing 'digits/90' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/million' (language 'en')
-- Playing 'digits/8' (language 'en')
-- Playing 'digits/hundred' (language 'en')
-- Playing 'digits/30' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/thousand' (language 'en')
-- Playing 'digits/4' (language 'en')
-- Playing 'digits/hundred' (language 'en')
-- Playing 'digits/60' (language 'en')
-- Playing 'digits/5' (language 'en')
-- AGI Script agi-test.agi completed, returning 0
-- Executing Hangup("SIP/201-4415", "") in new stack
== Spawn extension (default, 290, 3) exited non-zero on 'SIP/201-4415'
I don't get the other stuff - eg the send file, send text, etc. I have an
Asterisk console open (used asterisk -r) on a putty session on a PC
connected over network. There is no other asterisk console open.
Also when I dial on a a Snom 190 or a Sipura-841 I hear all the digits as
above correctly. But on a Polycom 300 I get to the digit 30 and it then
seems to stop playing the digits. But they of course appear on the console.
Why am I not getting the send file stuff etc on the console? The Polycom
bit I expect is some setting on the phone I need to troubleshoot. But not
getting all the expected output from the agi script seems strange.
Is there possibly some problem with my environment? My handset? I am
running on Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j
Angus
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[Asterisk-Users] Best way to create IVR/voicemail system
Hello I want to setup a system where people can dial a number and then a system will ask them questions for which they will leave answers. Eg something like this: Answer Playback(whatisyournamemsg) Record(yourname:gsm) Playback(whatisyourheight) Record(yourheight:gsm) Playback(thankyou) Hangup Is this the best way to do this sort of thing? Do users then just access the responses by eg *98 - or does this work a little differently to voicemail? How do we retrieve the responses? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Swap between callers
Hello On business phones it is often possible to have call waiting (think that is the feature) whereby if you are talking to a caller you can see another caller has called and you can swap between callers. For example, to say hello, I am on call with someone else now can I call you back. How can this be implemented using SIP IP phones. Do you need to setup two or more lines? How is it done? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] strange wave like noise on sip handset
No it happens on our asterisk and at a customers. Not that noticeable but not crystal clear. Didn't happen on a Snom 190. I have been working my way through IP handsets with these results: Grandstream BT-100 series. OKish for the price but a bit echoy. Grandstream GXP-2000 - OK but if used on hands free a bit echoy. Snom 190. Very clear. However, on a customer site they complained that full volume was still not load enough. But didn't extensively test. Sipura SPA-841 - when receiving an incoming call echoy for about 2-3 seconds at start of call then echo went away. Remote end did not hear any echo. Also wave like hiss as per my message. Next phones to try are a Polycom 300 and a CISCO 7940. I suppose it depends on how demanding customer is. I would hope that I can find a phone with no echo / hiss /other problems. Perhaps I need to think about using channel banks/FXS cards and analog phones! But would prefer IP phones for flexibility etc. Anyone found a perfect IP phone? Angus - Original Message - From: "Leif Madsen" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, October 01, 2005 2:33 PM Subject: Re: [Asterisk-Users] strange wave like noise on sip handset On 9/30/05, Angus Comber <[EMAIL PROTECTED]> wrote: On a Sipura SPA-841 handset (and also at other end) you hear a sea wave like sound - it gets louder then softer and continually repeats. I don't remember hearing this when using other handsets. But what is this effect? How can I reduce it? I heard the same thing from a remote users Polycom 501 - seems it was sitting too close to a fan in a computer. Could it be something similar to that? Just a thought since this happened to me yesterday :) -- Leif Madsen - http://www.leifmadsen.com Astricon 2005, Anaheim, CA, October 12-14 http://www.astricon.net http://www.oreilly.com/catalog/asterisk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to create IVR system using *
Hello I want to setup a system where people can dial a number and then a system will ask them questions for which they will leave answers. Eg something like this: Answer Playback(whatisyournamemsg) Record(yourname:gsm) Playback(whatisyourheight) Record(yourheight:gsm) Playback(thankyou) Hangup Is this the best way to do this sort of thing? Do users then just access the responses by eg *98 - or does this work a little differently to voicemail? How do we retrieve the responses? Or can I email the responses as WAV files? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why does the s extension not work inmy extensions.conf file
But I thought s was start and so should not need to do this? Angus - Original Message - From: "Matt Riddell" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, September 30, 2005 1:55 PM Subject: Re: [Asterisk-Users] Why does the s extension not work inmy extensions.conf file Angus Comber wrote: Hello In my extensions.conf file: [frompstnisdn] exten => s,1,Dial(SIP/200&SIP/202,20) exten => s,2,Voicemail(su200) exten => s,3,Hangup If you really want to use s, you will need to add an extension: exten => 787367,1,Goto(s,1) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] strange wave like noise on sip handset
Hello On a Sipura SPA-841 handset (and also at other end) you hear a sea wave like sound - it gets louder then softer and continually repeats. I don't remember hearing this when using other handsets. But what is this effect? How can I reduce it? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and telephone volume
Hello I am using a Snom 190 and the quality seems OK. Trouble is the volume is quite low and full volume on the Snom is still too low. Is there something I can do on the asterisk to increase the volume? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] analog phone/door buzzer going through a SipuraSPA2000 ATA dials really slowly
The unit dials 300 and in my extensions.conf I have: exten => 300,1,Dial(SIP/200&SIP/201,30) exten => 300,2,Hangup So perhaps it is some setting in the Sipura ATA? - Original Message - From: "Matt Riddell" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, September 30, 2005 1:51 PM Subject: Re: [Asterisk-Users] analog phone/door buzzer going through a SipuraSPA2000 ATA dials really slowly Angus Comber wrote: Hello We have setup a doorbell which has an inbuilt analog phone which is connected to our Asterisk via a SPA2000 ATA. The problem we are getting is that when a caller presses the buzzer it is taking two or more minutes to finally call the reception phone. Asterisk will not cause it to wait two or more minutes. 3 seconds yes, 2 minutes, no... Unless you have some funky gotoifs or loops or waits etc.. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] analog phone/door buzzer going through a Sipura SPA2000 ATA dials really slowly
Hello We have setup a doorbell which has an inbuilt analog phone which is connected to our Asterisk via a SPA2000 ATA. The problem we are getting is that when a caller presses the buzzer it is taking two or more minutes to finally call the reception phone. In the SPA2000 I have set dtmfmode to be inband. I notice that with the asterisk you dial a number and then it waits for a timeout before dialing number. I think you use a # to say - just dial now. Well we can't program a # into the door system, but could program in another character. Is it possible to use another character? Any ideas would be much appreciated. Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Will a VIA Epia ME6000 with a 600MHz Eden fanless CPU be suffiecient for 8 extension system?
Hello I am using a VIA Epia ME6000 with a 600MHz Eden Fanless CPU. Is this likely to be enough power for a 8 extension system with 6 external pstn lines? How important is cpu? Is there some measure, eg xMHz CPU per extension or something benchmark? I have installed 512MB memory - again any benchmark for asterisk memory usage? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel TDM questions
I think the Asterisk must answer the call to be able to then dial out on the second port. This is what happens on any other PBX I have worked with in this sort of scenario. Is this a problem for you? Angus - Original Message - From: Chee Foong To: asterisk-users@lists.digium.com Sent: Friday, September 30, 2005 10:20 AM Subject: [Asterisk-Users] Zaptel TDM questions Hello, I have a TDM04B. I make a call into the first port of the card. Once asterisk receive the call, it will make another call out using the second port. From what i have observerd as soon as the called party on the second port starts ringing asterisk show the following : -- Zap/2-1 answered Zap/1-1 Any idea why asterisk thinks the call has been answered while actually the phone is still ringing? Anybody know how to avoid asterisk to answer the call while ringing? Also, I have no Answer or any Playback command in the dial plan before making a call out of second port. I have also try setting callprogress to yes/no but the results are the same. Thanks CCF ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why does the s extension not work in my extensions.conf file
Hello In my extensions.conf file: [frompstnisdn] exten => s,1,Dial(SIP/200&SIP/202,20) exten => s,2,Voicemail(su200) exten => s,3,Hangup I use the s, start, extension to handle incoming calls. In my zapata.conf: context=frompstnisdn This works ok on another asterisk box I setup. But on incoming calls I get: -- Extension '787367' in context 'frompstnisdn' from '07768385144' does not exist. Rejecting call on channel 0/1, span 1 -- Saved useragent "X-Lite release 1103m" for peer 202 -- Extension '787367' in context 'frompstnisdn' from '07768385144' does not exist. Rejecting call on channel 0/1, span 1 Do I need to enable something to be able to use the s in extensions.conf? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot figure out why calls from myAsteriskappear to be fr
I use a BT ISDN line. But calls go through Onetel. Bizarre, this behaviour has now stopped. Country code now no longer part of CLI seen on my mobile. Perhaps it is as you say, something my least cost routing company, Onetel are doing! Angus - Original Message - From: "David J Carter" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Thursday, September 29, 2005 5:52 PM Subject: RE: [Asterisk-Users] Cannot figure out why calls from myAsteriskappear to be fr Do you use BT for you outgoing calls? Or are you using another provider? I have one customer who uses another provider and there calls come to me with some strange CLI numbers. It seems to be they break out where the best rates are at that time. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ian Bonham Sent: 29 September 2005 15:59 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Cannot figure out why calls from myAsteriskappear to be fr Not sure about the Digium, but I can tell you +34 is Spain, if that helps you track anything down? I assume you've tested the line with a normal phone to make sure it's not a telco fault? Ian From: "Angus Comber" <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion To: Subject: [Asterisk-Users] Cannot figure out why calls from my Asteriskappear to be from country code +34? Date: Thu, 29 Sep 2005 15:32:39 +0100 Hello When I dial out from my Asterisk (using Digium analog TDM04B card over pstn line), calls appear to be from +34 I am in UK which is +44 so cannot work out why seeing +34. In my zapata.conf I have: loadzone = uk defaultzone = uk I can't find any country specific stuff in any other conf files. Any ideas how I can correctly set so that calls from my asterisk do not have +34? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot figure out why calls from myAsteriskappear to be fr
I am seeing this by calling my Nokia mobile phone - using Vodafone in UK. If I substitute Asterisk for an Avaya IP Office then just get: 020 8878 7367 - ie my number but without the country code. So it must be something that the Asterisk is doing. Angus - Original Message - From: "Ian Bonham" <[EMAIL PROTECTED]> To: Sent: Thursday, September 29, 2005 3:59 PM Subject: RE: [Asterisk-Users] Cannot figure out why calls from myAsteriskappear to be fr Not sure about the Digium, but I can tell you +34 is Spain, if that helps you track anything down? I assume you've tested the line with a normal phone to make sure it's not a telco fault? Ian From: "Angus Comber" <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion To: Subject: [Asterisk-Users] Cannot figure out why calls from my Asteriskappear to be from country code +34? Date: Thu, 29 Sep 2005 15:32:39 +0100 Hello When I dial out from my Asterisk (using Digium analog TDM04B card over pstn line), calls appear to be from +34 I am in UK which is +44 so cannot work out why seeing +34. In my zapata.conf I have: loadzone = uk defaultzone = uk I can't find any country specific stuff in any other conf files. Any ideas how I can correctly set so that calls from my asterisk do not have +34? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cannot figure out why calls from my Asterisk appear to be from country code +34?
Hello When I dial out from my Asterisk (using Digium analog TDM04B card over pstn line), calls appear to be from +34 I am in UK which is +44 so cannot work out why seeing +34. In my zapata.conf I have: loadzone = uk defaultzone = uk I can't find any country specific stuff in any other conf files. Any ideas how I can correctly set so that calls from my asterisk do not have +34? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk overheating on VIA Epia M Seriesmotherboard
But the systems are sold in this configuration. There is a fan option. I chose the fanless option. Angus - Original Message - From: "C F" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, September 06, 2005 1:28 AM Subject: Re: [Asterisk-Users] Asterisk overheating on VIA Epia M Seriesmotherboard As you suspected, the problem is the fact that you don't have a fan. Since a machine that runs just a file server does not require much CPU power, the CPU doesn't get too hot. However Asterisk does use lots of CPU, therefore the CPU is hot, and yes the problem of stopping to work is because of the CPU being overheated, you are lucky that the computer booted after that, in most cases the overheating of a CPU means that the CPU expanded too much, when you shut it down it cools off, and shrinks, which could result in cracking the CPU. You should never run a CPU without it's fan if it's meant to run with a fan. Even if running it just as a file server. The fact that you are lucky doesn't mean that you don't need a fan. On 9/5/05, Angus Comber <[EMAIL PROTECTED]> wrote: Hello I am running Asterisk on SUSE Linux Professional 9.3 on a VIA Epia M Series motherboard - CPU runs at 1GHz. There is no fan - just a large heatsink. Currently system is running off standard IDE hard drive - because I couldn't get astlinux to run with my Digium TDM04B card (only PCI card in system). Strangely I also have the same system also running SUSE Linux running as a file server and that does not run so hot and does not overheat? Why the difference? Just booting up both systems for 15 minutes you can tell the Asterisk box is quite a bit hotter. Also the Asterisk box overheated (well think that was the problem) and stopped operating as PBX at one stage. Anyone any experience of this sort of thing? any ideas how to fix - ideally I don't want to have to fit a fan. Is SUSE not the best distro to use for this sort of thing? Should it be something to take up with VIA? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk overheating on VIA Epia M Series motherboard
Hello I am running Asterisk on SUSE Linux Professional 9.3 on a VIA Epia M Series motherboard - CPU runs at 1GHz. There is no fan - just a large heatsink. Currently system is running off standard IDE hard drive - because I couldn't get astlinux to run with my Digium TDM04B card (only PCI card in system). Strangely I also have the same system also running SUSE Linux running as a file server and that does not run so hot and does not overheat? Why the difference? Just booting up both systems for 15 minutes you can tell the Asterisk box is quite a bit hotter. Also the Asterisk box overheated (well think that was the problem) and stopped operating as PBX at one stage. Anyone any experience of this sort of thing? any ideas how to fix - ideally I don't want to have to fit a fan. Is SUSE not the best distro to use for this sort of thing? Should it be something to take up with VIA? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why do I get pbx.c 1645 pbx_extension_helper: No application 'Voicemailman' for extension
Does VoicemailMan have to be installed ? Why not available. I have setup a
mailbox in voicemail.conf and I can leave a voicemail - just cannot pickup
up using *97.
My *97 code in extensions.conf:
exten => *97,1,Answer
exten => *97,2,VoicemailMain([EMAIL PROTECTED])
exten => *97,3,Hangup
asterisk console:
Verbosity was 8 and is now 12
-- Executing Answer("SIP/200-d83a", "") in new stack
Aug 20 18:57:45 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No
application 'VoicemailMan' for extension (default, *97, 2)
== Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-d83a'
-- Executing Answer("SIP/200-81f6", "") in new stack
Aug 20 18:57:59 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No
application 'VoicemailMan' for extension (default, *97, 2)
== Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-81f6'
-- Executing Answer("SIP/201-a86c", "") in new stack
Aug 20 19:00:24 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No
application 'VoicemailMan' for extension (default, *97, 2)
== Spawn extension (default, *97, 2) exited non-zero on 'SIP/201-a86c'
-- Executing Dial("SIP/201-1e08", "SIP/200|20|Ttm") in new stack
-- Called 200
-- Started music on hold, class 'default', on SIP/201-1e08
-- SIP/200-b925 is ringing
-- Stopped music on hold on SIP/201-1e08
-- Nobody picked up in 2 ms
-- Executing VoiceMail("SIP/201-1e08", "su200") in new stack
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/200/INBOX/msg format: wav49,
0x818eb40
-- x=1, open writing:
/var/spool/asterisk/voicemail/default/200/INBOX/msg format: gsm,
0x813a7e8
-- x=2, open writing:
/var/spool/asterisk/voicemail/default/200/INBOX/msg format: wav,
0x818ed88
-- User hung up
== Spawn extension (default, 200, 2) exited non-zero on 'SIP/201-1e08'
-- Executing Answer("SIP/200-4b1a", "") in new stack
Aug 20 19:01:57 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No
application 'VoicemailMan' for extension (default, *97, 2)
== Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-4b1a'
-- Executing Answer("SIP/200-5369", "") in new stack
Aug 20 19:02:11 WARNING[6014]: pbx.c:1645 pbx_extension_helper: No
application 'VoicemailMan' for extension (default, *97, 2)
== Spawn extension (default, *97, 2) exited non-zero on 'SIP/200-5369'
linux*CLI>
Angus
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[Asterisk-Users] Is there a way in dialplan to determine if call is incoming or outgoing if callerid presentation not enabled on line?
Hello If callerid is not available on an external line, how can you tell if call is incoming or outgoing? Angus ___ 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
Re: [Asterisk-Users] asterisk seems to load but cannot connectusing-r?
It was my own stupid fault for installing the asterisk version available in the SUSE distribution and then downloading and installing the latest version. Another thing not to do! Uninstalled old and re-installed asterisk and it worked! Angus - Original Message - From: "Angus Comber" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, August 19, 2005 8:58 AM Subject: Re: [Asterisk-Users] asterisk seems to load but cannot connectusing-r? But when I load Asterisk it doesn't complain. Get 2 warnings: [chan_capi.so]Aug 18 20:43:32 WARNING[13248]: loader.c:314 __load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed Aug 18 20:43:32 WARNING[13248]: loader.c:543 load_modules: Loading module chan_capi.so failed! So Asterisk must be crashing after starting? What do I do now? If I look in /var/log/asterisk see this only: Aug 18 21:47:00 WARNING[6079] loader.c: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed Aug 18 21:47:00 WARNING[6079] loader.c: Loading module chan_capi.so failed! Aug 19 08:48:12 NOTICE[8271] cdr.c: CDR simple logging enabled. Aug 19 08:48:12 WARNING[8271] loader.c: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed Aug 19 08:48:12 WARNING[8271] loader.c: Loading module chan_capi.so failed! linux:/var/log/asterisk # Angus - Original Message - From: "Dave Cotton" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, August 19, 2005 8:22 AM Subject: Re: [Asterisk-Users] asterisk seems to load but cannot connectusing-r? On Fri, 2005-08-19 at 08:08 +0100, Angus Comber wrote: Still get same: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) The error message says it all. It thinks it's not running. Check with the ps command. -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk seems to load but cannot connectusing-r?
But when I load Asterisk it doesn't complain. Get 2 warnings: [chan_capi.so]Aug 18 20:43:32 WARNING[13248]: loader.c:314 __load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed Aug 18 20:43:32 WARNING[13248]: loader.c:543 load_modules: Loading module chan_capi.so failed! So Asterisk must be crashing after starting? What do I do now? If I look in /var/log/asterisk see this only: Aug 18 21:47:00 WARNING[6079] loader.c: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed Aug 18 21:47:00 WARNING[6079] loader.c: Loading module chan_capi.so failed! Aug 19 08:48:12 NOTICE[8271] cdr.c: CDR simple logging enabled. Aug 19 08:48:12 WARNING[8271] loader.c: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed Aug 19 08:48:12 WARNING[8271] loader.c: Loading module chan_capi.so failed! linux:/var/log/asterisk # Angus - Original Message - From: "Dave Cotton" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, August 19, 2005 8:22 AM Subject: Re: [Asterisk-Users] asterisk seems to load but cannot connectusing-r? On Fri, 2005-08-19 at 08:08 +0100, Angus Comber wrote: Still get same: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) The error message says it all. It thinks it's not running. Check with the ps command. -- Dave Cotton <[EMAIL PROTECTED]> ___ 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
Re: [Asterisk-Users] asterisk seems to load but cannot connect using-r?
Still get same: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) Angus - Original Message - From: "Fábio Sakai" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, August 18, 2005 9:18 PM Subject: RES: [Asterisk-Users] asterisk seems to load but cannot connect using-r? Angus, Try this command: asterisk -c -r Fábio Sakai DGX - Digital Express Suporte CosmoCall [EMAIL PROTECTED] +55 11 3049.8109 -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Angus Comber Enviada em: quinta-feira, 18 de agosto de 2005 16:58 Para: asterisk-users@lists.digium.com Assunto: [Asterisk-Users] asterisk seems to load but cannot connect using -r? I installed asterisk on SUSE 9.3. Stupidly I loaded selected to load asterisk from the SUSE DVD - then installed latest asterisk head using cvs. At end of asterisk compilation mentioned modules in /modules where from another installation. My telephony cards working ok and if run asterisk just get these warnings: [chan_capi.so]Aug 18 20:43:32 WARNING[13248]: loader.c:314 __load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed Aug 18 20:43:32 WARNING[13248]: loader.c:543 load_modules: Loading module chan_capi.so failed! Are they serious? Then I try: linux:/var/run/asterisk # asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) linux:/var/run/asterisk # ls -al total 5 drwxr-x--- 2 asterisk root 112 Aug 18 20:43 . drwxr-xr-x 13 root root 880 Aug 18 18:44 .. srwxr-xr-x 1 root root 0 Aug 18 20:43 asterisk.ctl -rw-r--r-- 1 root root 6 Aug 18 20:43 asterisk.pid linux:/var/run/asterisk # but /var/run/asterisk/asterisk.ctl does exit? how can I fix this? Is it a problem with those modules in /usr/lib/asterisk/modules? Should I delete them? What? Angus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk seems to load but cannot connect using -r ?
I installed asterisk on SUSE 9.3. Stupidly I loaded selected to load asterisk from the SUSE DVD - then installed latest asterisk head using cvs. At end of asterisk compilation mentioned modules in /modules where from another installation. My telephony cards working ok and if run asterisk just get these warnings: [chan_capi.so]Aug 18 20:43:32 WARNING[13248]: loader.c:314 __load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: ast_smoother_feed Aug 18 20:43:32 WARNING[13248]: loader.c:543 load_modules: Loading module chan_capi.so failed! Are they serious? Then I try: linux:/var/run/asterisk # asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) linux:/var/run/asterisk # ls -al total 5 drwxr-x--- 2 asterisk root 112 Aug 18 20:43 . drwxr-xr-x 13 root root 880 Aug 18 18:44 .. srwxr-xr-x 1 root root 0 Aug 18 20:43 asterisk.ctl -rw-r--r-- 1 root root 6 Aug 18 20:43 asterisk.pid linux:/var/run/asterisk # but /var/run/asterisk/asterisk.ctl does exit? how can I fix this? Is it a problem with those modules in /usr/lib/asterisk/modules? Should I delete them? What? Angus ___ 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] do not appear to have the sources for the 2.6.11.4-20a-default kernel installed
When I attempt to compile the zaptel driver (latest CVS HEAD) I get this compile error: You do not appear to have the sources for the 2.6.11.4-20a-default kernel installed. make: *** [linux26] Error 1 When I load YaST I see kernel-source 2.6.11.4 as installed version. So why do I get this message? YaST is just a SUSE config menu program. What can I do? I am running SUSE Pro 9.3 and have today updated using YaST. Angus ___ 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
Re: [Asterisk-Users] Does intel 865 board works fine with Asterisk
I have one Asterisk system working with a Junghanns BRI card and another working with a Digium TDM card with an Intel D865 motherboard. Angus - Original Message - From: "jonny hashem" <[EMAIL PROTECTED]> To: Sent: Wednesday, August 17, 2005 6:14 PM Subject: [Asterisk-Users] Does intel 865 board works fine with Asterisk Hi: I would like to know what are the issues I need to look for in a chipset board so I can make sure it works fine with digium cards and Asterisk . Is intel board 865 fits the description? Regards; __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Automatic start with SuSe linux
You could just add the line asterisk to /etc/init.d/boot.local Angus - Original Message - From: <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, August 17, 2005 11:27 AM Subject: [Asterisk-Users] Automatic start with SuSe linux Hi! I'm trying to start asterisk at boottime. Since SuSe has no rc.local like in Redhat linux, I need asterisk starting script to /etc/init.d/rc3.d -directory (I assume it is like that if i want automated asterisk startup). Do you have any experience how this is implemented in SuSe, and if you have some useful script for starting asterisk, I would be very, i mean VERY pleased? Thank you all in advance! This mail sent through L-secure: http://www.l-secure.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What should my next steps in troubleshooting this TDM04B error be?
Hello I have installed a TDM04B and disabled any devices not required in my PC. (TDM04B is analog card with 4 ports to plug into telephone co lines). I am running this version of * Asterisk CVS-HEAD-01/10/05-02:11:15-AstLinux built by [EMAIL PROTECTED] on a i686 running Linux As you see below the wctdm module is loaded: pbx root # lsmod Module Size Used by binfmt_misc 12296 1 - Live 0xde839000 wctdm 129216 0 - Live 0xde855000 zaptel 235844 1 wctdm, Live 0xde877000 hdlc 24576 1 zaptel, Live 0xde84e000 syncppp 17116 1 hdlc, Live 0xde848000 ppp_generic 30612 1 zaptel, Live 0xde83f000 slhc 7808 1 ppp_generic, Live 0xde829000 crc_ccitt 2432 1 zaptel, Live 0xde806000 via_rhine 21252 0 - Live 0xde82d000 mii 5120 1 via_rhine, Live 0xde81d000 crc32 4608 1 via_rhine, Live 0xde81a000 rtc 12748 0 - Live 0xde82 But running ztfcg gives me this error: pbx root # ztcfg -v Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) What does this mean exactly? Is it saying it can't find the hardware? Then get these errors loading Asterisk: == Parsing '/etc/asterisk/zapata.conf': Found Aug 16 19:56:12 WARNING[363]: chan_zap.c:792 zt_open: Unable to specify channel 1: No such device or address Aug 16 19:56:12 ERROR[363]: chan_zap.c:6327 mkintf: Unable to open channel 1: No such device or address here = 0, tmp->channel = 1, channel = 1 Aug 16 19:56:12 ERROR[363]: chan_zap.c:9337 setup_zap: Unable to register channel '1' Aug 16 19:56:12 WARNING[363]: loader.c:396 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Zap' Aug 16 19:56:12 WARNING[363]: loader.c:501 load_modules: Loading module chan_zap.so failed! My zaptel.conf: fxsks=1-4 loadzone = uk defaultzone = uk My zapata.conf (abbreviated): [channels] context=default group=1 signalling=fxs_ks channel => 1-4 What do I need to look at next? Angus ___ 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
Re: [Asterisk-Users] Module wcfxs - is it not part of astlinux?
I am running astlinux 0.2.8 - ie latest latest version. OK so wctdm is alias to same as wcfxs. But even if I load that, it loads OK pbx sbin # lsmod Module Size Used by binfmt_misc 12296 1 - Live 0xde839000 wctdm 129216 0 - Live 0xde855000 zaptel 235844 1 wctdm, Live 0xde877000 hdlc 24576 1 zaptel, Live 0xde84e000 syncppp 17116 1 hdlc, Live 0xde848000 ppp_generic 30612 1 zaptel, Live 0xde83f000 slhc 7808 1 ppp_generic, Live 0xde829000 crc_ccitt 2432 1 zaptel, Live 0xde806000 via_rhine 21252 0 - Live 0xde82d000 mii 5120 1 via_rhine, Live 0xde81d000 crc32 4608 1 via_rhine, Live 0xde81a000 rtc 12748 0 - Live 0xde82 pbx sbin # But don't see it with a cat /proc/interrupts (I have disabled all unused devices in BIOS). pbx sbin # cat /proc/interrupts CPU0 0:1348003 XT-PIC timer 1: 8 XT-PIC i8042 2: 0 XT-PIC cascade 5: 0 XT-PIC uhci_hcd 8: 0 XT-PIC rtc 9: 0 XT-PIC acpi 10: 0 XT-PIC uhci_hcd 11: 1226 XT-PIC uhci_hcd, eth0 12: 0 XT-PIC ehci_hcd 14: 13563 XT-PIC ide0 15: 71 XT-PIC ide1 NMI: 0 LOC: 0 ERR: 0 MIS: 0 pbx sbin # And still can't load Asterisk and ztcfg gives me pbx sbin # ztcfg -v Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) The card was working fine in other PC - so I have to assume it is some config issue. Angus - Original Message - From: "Tzafrir Cohen" <[EMAIL PROTECTED]> To: Sent: Sunday, August 14, 2005 11:45 AM Subject: Re: [Asterisk-Users] Module wcfxs - is it not part of astlinux? On Sun, Aug 14, 2005 at 11:37:00AM +0100, Angus Comber wrote: Hello I am (attempting) to run the astlinux version Which version? of Asterisk on a VIA embedded platform. I have a TDM04B and pretty sure zaptel.conf and zapata.conf setup OK. They worked fine with same card in traditional PC anyway. I think need the module wcfxs for a Digium TDM04B card. Is this module not part of astlinux? Do I need to download it? Or is it in opt? I see wctdm - but think that is for X100 card. X100*P* uses wcfxo . IIRC in 1.0 TDM04B still uses wcfxs and wcfxo, depending on the module, but maybe just wcfxs. In later 1.0 zaptels, wctdm is an alias to wcfxs. In HEAD/1.2 wcfxs is gone and replaced with wctdm. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Module wcfxs - is it not part of astlinux?
Hello I am (attempting) to run the astlinux version of Asterisk on a VIA embedded platform. I have a TDM04B and pretty sure zaptel.conf and zapata.conf setup OK. They worked fine with same card in traditional PC anyway. I think need the module wcfxs for a Digium TDM04B card. Is this module not part of astlinux? Do I need to download it? Or is it in opt? I see wctdm - but think that is for X100 card. in rc.conf I have the entry ZAPMODS="wcfxs" in ## Better Zaptel support ## section. I commented out ZAPMODS="wctdm" because I thought that was for X100 card. (rc.conf is a specific astlinux central config file). I get these errors: Aug 14 10:18:39 pbx local0.warn asterisk[297]: WARNING[297]: loader.c:396 in ast_load_resource: chan_zap.so: load_module failed, returning -1 Aug 14 10:18:39 pbx local0.warn asterisk[297]: WARNING[297]: loader.c:501 in load_modules: Loading module chan_zap.so failed! Presumably I just need to find the wcfxs module? Angus p.s. I realise there is an astlinux mailing list - just trying to get max coverage as need to get this working asap ;) ___ 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
Re: [Asterisk-Users] Can call from iax extn but cannot call it -unable to cteate channel iax
I have it working now thanks. I should have used IAX2 as the technology
rather than IAX in extensions.conf I have also donned string vest and will
promise to spend more time in wiki.
Angus
- Original Message -
From: "Gurminder Arora" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, August 08, 2005 8:22 AM
Subject: Re: [Asterisk-Users] Can call from iax extn but cannot call
it -unable to cteate channel iax
Hi Angus,
If you can send your general settings of iax.conf may be
I can work it out.
Regards
Gurminder
On 8/7/05, Angus Comber <[EMAIL PROTECTED]> wrote:
Hello
I have created an iax exten in my iax.conf file:
[300]
type=friend
username=300
secret=***
context=default
host=dynamic
callerid="some name" <300>
auth=md5
Then in my extensions.conf I have:
exten => 300,1,Dial(IAX/${EXTEN},20)
exten => 300,2,Hangup
I can dial from iaxComm (a soft IAX client) and that works fine. But when
I
try to dial 300 get:
WARNING[22077]: channel.c1970 ast_request: No channel type registered for
'IAX'
NOTICE[22077]: app_dial.c:777 dial_exec: Unable to create channel of type
'IAX'
I have restarted Asterisk after config change.
What have I not done. I am just testing the iaxComm program.
Angus
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[Asterisk-Users] Can call from iax extn but cannot call it - unable to cteate channel iax
Hello
I have created an iax exten in my iax.conf file:
[300]
type=friend
username=300
secret=***
context=default
host=dynamic
callerid="some name" <300>
auth=md5
Then in my extensions.conf I have:
exten => 300,1,Dial(IAX/${EXTEN},20)
exten => 300,2,Hangup
I can dial from iaxComm (a soft IAX client) and that works fine. But when I
try to dial 300 get:
WARNING[22077]: channel.c1970 ast_request: No channel type registered for
'IAX'
NOTICE[22077]: app_dial.c:777 dial_exec: Unable to create channel of type
'IAX'
I have restarted Asterisk after config change.
What have I not done. I am just testing the iaxComm program.
Angus
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Re: [Asterisk-Users] Is this echo problem down to IP Phone hardware?
- Original Message - From: "Steve Underwood" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, August 05, 2005 7:39 PM Subject: Re: [Asterisk-Users] Is this echo problem down to IP Phone hardware? Kris Boutilier wrote: This known as is 'acoustic echo' or 'room reverb' and involves mathematics that is quite a bit different from that used when cancelling regular 'reflected electrical signal' echos, as the signal is being acousically distorted as it echos around the room. On many handsfree handsets it doesn't manifest itself until you move into a physically large room, which increases the reflection delay and overwhelms the internal mechanisms. The maths is exactly the same. However, it is certainly true that a lot of acoustic echo cancellers don't deal with long enough echoes to be effective in large spaces. It would need to be handled internally by the handset or you would need to insert a hardware echo canceller capable of dealing with this type of echo, assuming your signal is exposed on a T1 somewhere. If it's IP all the way for you then you're really just down to the handset vendors as far as I know - Asterisk doesn't currently offer any form of echo cancellation on the VoIP side. In the IP world the echo must be killed by the phone itself. You cannot echo cancel on the IP side of a switch like Asterisk. The echo path length needs to be constant for any known echo cancellation process to work. IP path lengths are not constant. Hello I have a Grandstream GXP2000 with latest firmware. When I use it holding the handpiece I don't hear any echo - neither does other end. However, if I use it handsfree, the other end notices echo when they speak - ie their voice is echoy. I hear their voice being a bit echoy. The Grandstreams are much maligned, but they actually do a better job in this area than most products. As said above, if you are using this in a large space the echo canceller in the phone may not cancel a long enough echo to be very effective. If it fails to kill the echo in a small room something is wrong. * The room is 15 foot by 22 foot. Not massive. When you say something is wrong, what should I be looking at? I will buy a Cisco 7940 as suggested previously to see if the handset does make a difference. In my sip.conf I allow ulaw, alaw, g723.1, g729 and gsm. Should I tighten this down to fewer? Which ones? More? Is this purely down to the IP Phone? Is there anything I can do about it? I considered buying a more expensive phone - eg a Snom to see what they were like for echo. Is there something I can do with the Asterisk? codec to use? Anything? A snom might be a poor choice. People tell me they don't even echo cancel the handset. If a hard of hearing user turns up the handset volume the caller hears considerable echo. * Thanks. I will test with a Cisco. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is this echo problem down to IP Phone hardware?
Hello I have a Grandstream GXP2000 with latest firmware. When I use it holding the handpiece I don't hear any echo - neither does other end. However, if I use it handsfree, the other end notices echo when they speak - ie their voice is echoy. I hear their voice being a bit echoy. Is this purely down to the IP Phone? Is there anything I can do about it? I considered buying a more expensive phone - eg a Snom to see what they were like for echo. Is there something I can do with the Asterisk? codec to use? Anything? Angus ___ 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] Asterisk TDM card connected to phone lines AND fax line
Hello I want to setup an Asterisk with three analog lines. Two of the analog lines are the main office number. The other line is the fax number. The fax machine plugs into the line 3 but also will be a connection to the third port on a Digium analog card. Reason for the third line into Asterisk is so if two lines in use someone can still dial out over third (fax) line. Is this going to cause a problem? How would I stop the Idiom card answering on line 3? Angus ___ 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] Anyone know of an open source sip video phone like eyebeam available?
I just wondered - might save me some development effort! Angus ___ 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] SIP phone procedural question
Hello A lot of my customers have people who are in the office most of the time but occasionally wish to work from home. So they may have a sip phone which is extension 208 in the office. When they work from home they can of course plug in a sip phone into their broadband connection and work with that. But it would be ideal if they could be same extension as phone in office. If they try to register as same sip user - eg extn 208 - will it work. Then problem is phone on their desk will still ring p***ing all their colleagues off. How do people deal with this sort of thing. Ideally, would want person to be able to easily switch from office to home but use same extension. Or does sip somehow deal with this? Is there a standard sip way of dealing with this? Angus ___ 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] Is there an upper extension limit to Asterisk?
Hello I have an application for Asterisk which could involve potentially 5000 or more extensions. Possibly this number of people making calls. All calls would be internal. Could enough hardware be thrown at the problem to make this work? Anyone setup an installation of this size? Any comments on how to size it, etc? Angus ___ 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] Auto loading of qozap module
Apologies for being a bit of a Linux newbie... I have got a working * system but each time I reboot my box I need to: modprobe qozap ztcfg asterisk Now I realise this is really a Linux question but I am struggling with the problem and any help would be much appreciated. There is a module qozap.ko - which if I do a find I see in /lib/modules/2.6.11.4-11.4.21.7-default/misc/qozap.ko Is this the module? If it is here, then why do I need to modprobe qozap? I have looked at /etc/init.d/rc - but this seems to be all about services! Wrong place to look? So somehow how do I load this module so it runs at startup? Angus ___ 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
Re: [Asterisk-Users] Should this work?
I tried your way (also sorted out two priority 2's
- actually a typo in my emails) - but same problem.
It seems to be my * does not like Zap\1 - if I zap
show channels I get
linux*CLI> zap show channels
Chan Extension Context
Language
MusicOnHold pseudo
default
en
1
default
en
2
default
en
4
default
en
5
default
en
7
default
en
8
default
en
10
default
en
11
default en
And zap show channel 1 I get:
linux*CLI> zap show channel 1Channel:
1File Descriptor: 11Span: 1Extension:Dialing: noContext:
defaultCaller ID string:Destroy: 0InAlarm: 1Signalling Type: PRI
SignallingOwner: Real: Callwait:
Threeway: Confno: -1Propagated Conference:
-1Real in conference: 0DSP: noRelax DTMF: noDialing/CallwaitCAS:
0/0Default law: alawFax Handled: noPulse phone: noEcho
Cancellation: 128 taps unless TDM bridged, currently OFFPRI Flags:PRI
Logical Span: ImplicitActual Hookstate:
Onhooklinux*CLI>
Is it I don't call it Zap but something else - as
it is ISDN BRI?
Angus
- Original Message -
From:
Jason
Walker
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, July 25, 2005 8:45 PM
Subject: Re: [Asterisk-Users] Should this
work?
Have you defined the context "default" in the
extensions.conf for outbound dialing in the globals section?
For example, I have my ZAP channels identified as
OUTBND1 not ZAP in the global section. This new global identifier is pointed
to ZAP/g1
[globals]
OUTBND1=Zap/g1
Instead of ZAP in my dial plan to call out, I use
${OUTBND1}.
Yours:
; for dialing outbound - over ISDN line - this
bit does not work
exten =>
_9XX.,2,Dial(ZAP/g1/${EXTEN},60)
exten => _9XX.,2,Hangup
Mine would look like
this
exten =>
_9XX.,1,Dial(${OUTBND1}/${EXTEN},##)
exten => _9XX.,2,Hangup
This helps me to keep track of inbound T1s and
outbound T1s.
Also, you have 2 (2) priorities listed in your
example. You can't really do this.
JASON WALKER
- Original Message -
From:
Angus
Comber
To: asterisk-users@lists.digium.com
Sent: Monday, July 25, 2005 8:11
AM
Subject: [Asterisk-Users] Should this
work?
Hello
I am using a Junghans quadBRI ISDN card and it
is loaded and working. In Asterisk if I connect to ISDN line it is
detected and tells me so.
In my zapata.conf I have
(abbreviated):
[channels]
switchtype=euroisdn
signalling = bri_cpe
context=default
group=1
channel => 1-2
;plus group 2 - 4
zaptel.conf:
loadzone=ukdefaultzone=uk# qozap span
definitions# most of the values should be bogus because we are not
really
zaptelspan=1,1,3,ccs,amispan=2,0,3,ccs,amispan=3,0,3,ccs,amispan=4,0,3,ccs,ami
bchan=1,2dchan=3bchan=4,5dchan=6bchan=7,8dchan=9bchan=10,11dchan=12
Then in extensions.conf I have:
[default]
; this below for internal extensions - works
OK
exten =>
_2XX,1,Dial(SIP/${EXTEN},20,Ttm)
; for dialing outbound - over ISDN line - this
bit does not work
exten =>
_9XX.,2,Dial(ZAP/g1/${EXTEN},60)
exten => _9XX.,2,Hangup
Error I get is:
-- Executing
Dial("SIP/200-e433", "ZAP/g1/902088787367|60") in new stackJul 25
11:56:33 NOTICE[6723]: app_dial.c:777 dial_exec: Unable to create channel of
type 'ZAP' == Everyone is busy/congested at this
time -- Executing Hangup("SIP/200-e433", "") in new
stack == Spawn extension (default, 902088787367, 2) exited
non-zero on 'SIP/200-e433'
I am dialing with sip phones. They work
if dialing extensions internally but not if try to dial outside - eg dial 9
followed by number.
What have I not done right?
Angus
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Re: [Asterisk-Users] Should this work?
I changed to:
exten =>
_X.,1,Dial(ZAP/1/${EXTEN},60)
exten => _X.,2,Hangup
But still didn't work.
even though could see channel with zap show
channels - saw a channel 1
Angus
- Original Message -
From:
Jason
Walker
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, July 25, 2005 8:45 PM
Subject: Re: [Asterisk-Users] Should this
work?
Have you defined the context "default" in the
extensions.conf for outbound dialing in the globals section?
For example, I have my ZAP channels identified as
OUTBND1 not ZAP in the global section. This new global identifier is pointed
to ZAP/g1
[globals]
OUTBND1=Zap/g1
Instead of ZAP in my dial plan to call out, I use
${OUTBND1}.
Yours:
; for dialing outbound - over ISDN line - this
bit does not work
exten =>
_9XX.,2,Dial(ZAP/g1/${EXTEN},60)
exten => _9XX.,2,Hangup
Mine would look like
this
exten =>
_9XX.,1,Dial(${OUTBND1}/${EXTEN},##)
exten => _9XX.,2,Hangup
This helps me to keep track of inbound T1s and
outbound T1s.
Also, you have 2 (2) priorities listed in your
example. You can't really do this.
JASON WALKER
- Original Message -
From:
Angus
Comber
To: asterisk-users@lists.digium.com
Sent: Monday, July 25, 2005 8:11
AM
Subject: [Asterisk-Users] Should this
work?
Hello
I am using a Junghans quadBRI ISDN card and it
is loaded and working. In Asterisk if I connect to ISDN line it is
detected and tells me so.
In my zapata.conf I have
(abbreviated):
[channels]
switchtype=euroisdn
signalling = bri_cpe
context=default
group=1
channel => 1-2
;plus group 2 - 4
zaptel.conf:
loadzone=ukdefaultzone=uk# qozap span
definitions# most of the values should be bogus because we are not
really
zaptelspan=1,1,3,ccs,amispan=2,0,3,ccs,amispan=3,0,3,ccs,amispan=4,0,3,ccs,ami
bchan=1,2dchan=3bchan=4,5dchan=6bchan=7,8dchan=9bchan=10,11dchan=12
Then in extensions.conf I have:
[default]
; this below for internal extensions - works
OK
exten =>
_2XX,1,Dial(SIP/${EXTEN},20,Ttm)
; for dialing outbound - over ISDN line - this
bit does not work
exten =>
_9XX.,2,Dial(ZAP/g1/${EXTEN},60)
exten => _9XX.,2,Hangup
Error I get is:
-- Executing
Dial("SIP/200-e433", "ZAP/g1/902088787367|60") in new stackJul 25
11:56:33 NOTICE[6723]: app_dial.c:777 dial_exec: Unable to create channel of
type 'ZAP' == Everyone is busy/congested at this
time -- Executing Hangup("SIP/200-e433", "") in new
stack == Spawn extension (default, 902088787367, 2) exited
non-zero on 'SIP/200-e433'
I am dialing with sip phones. They work
if dialing extensions internally but not if try to dial outside - eg dial 9
followed by number.
What have I not done right?
Angus
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[Asterisk-Users] Should this work?
Hello
I am using a Junghans quadBRI ISDN card and it is
loaded and working. In Asterisk if I connect to ISDN line it is detected
and tells me so.
In my zapata.conf I have
(abbreviated):
[channels]
switchtype=euroisdn
signalling = bri_cpe
context=default
group=1
channel => 1-2
;plus group 2 - 4
zaptel.conf:
loadzone=ukdefaultzone=uk# qozap span
definitions# most of the values should be bogus because we are not really
zaptelspan=1,1,3,ccs,amispan=2,0,3,ccs,amispan=3,0,3,ccs,amispan=4,0,3,ccs,ami
bchan=1,2dchan=3bchan=4,5dchan=6bchan=7,8dchan=9bchan=10,11dchan=12
Then in extensions.conf I have:
[default]
; this below for internal extensions - works
OK
exten =>
_2XX,1,Dial(SIP/${EXTEN},20,Ttm)
; for dialing outbound - over ISDN line - this bit
does not work
exten =>
_9XX.,2,Dial(ZAP/g1/${EXTEN},60)
exten => _9XX.,2,Hangup
Error I get is:
-- Executing
Dial("SIP/200-e433", "ZAP/g1/902088787367|60") in new stackJul 25 11:56:33
NOTICE[6723]: app_dial.c:777 dial_exec: Unable to create channel of type
'ZAP' == Everyone is busy/congested at this time
-- Executing Hangup("SIP/200-e433", "") in new stack == Spawn
extension (default, 902088787367, 2) exited non-zero on
'SIP/200-e433'
I am dialing with sip phones. They work if
dialing extensions internally but not if try to dial outside - eg dial 9
followed by number.
What have I not done right?
Angus
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[Asterisk-Users] Need to ztcfg every time I reboot *
Hello I am sure this is a very basic Linux question. But every time I reboot my * I need to modprobe and then ztcfg After doing this I can then run * without it complaining about not loading a channel. The module being loaded is qozap - a ISDN card. What do I need to do to make the ztcfg configuration persistent? Angus ___ 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
Re: [Asterisk-Users] Why can't sip/200 call sip/202
That was another problem - now fixed.
Thanks for all your help on extensions.conf
Angus
- Original Message -
From: "Angus Comber" <[EMAIL PROTECTED]>
To: "Mark Edwards" <[EMAIL PROTECTED]>; "Asterisk Users Mailing
List - Non-Commercial Discussion"
Sent: Sunday, July 24, 2005 11:30 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
Sorry to be a pain... but
I restarted my * and now when I launch * get this:
== Parsing '/etc/asterisk/zapata.conf': Found
Jul 24 18:52:45 WARNING[6817]: chan_zap.c:932 zt_open: Unable to specify
channel 1: No such device or address
Jul 24 18:52:45 ERROR[6817]: chan_zap.c:6473 mkintf: Unable to open
channel 1: No such device or address
here = 0, tmp->channel = 1, channel = 1
Jul 24 18:52:45 ERROR[6817]: chan_zap.c:10303 setup_zap: Unable to
register channel '1-2'
Jul 24 18:52:45 WARNING[6817]: loader.c:345 ast_load_resource:
chan_zap.so: load_module failed, returning -1
== Unregistered channel type 'Tor'
== Unregistered channel type 'Zap'
Jul 24 18:52:45 WARNING[6817]: loader.c:440 load_modules: Loading module
chan_zap.so failed!
linux:~ # Ouch ... error while writing audio data: : Broken pipe
I have a Junghanns quadBRI card installed. I have modprobe qozap - so it
is loaded and seems to be working OK. I assume there was something in
extensions.conf which was somehow required. something to do with
[channels] ?
The error in chan_zap.c seems to be saying that channel 1 cannot be
opened.
Angus
- Original Message -
From: "Mark Edwards" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, July 24, 2005 10:13 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
OK Angus
just start here
mv extensions.conf extensions.conf.old
and create a new extensions.conf
[default]
exten => _2XX,1,Dial(SIP/${EXTEN},20,Ttm)
exten => _2XX,2,Hangup
just those 3 lines
do an 'extensions reload' in the CLI or just restart Asterisk
and see if it works
regards,
Mark.
On 7/25/05, Angus Comber <[EMAIL PROTECTED]> wrote:
I think the 777 may be a bit of a Red Herring. I dialed 777 as a test.
I
can't dial 202 from 200 if I actually dial 202!
My extensions.conf file:
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
; You can include other config files, use the #include command (without
the
';')
; Note that this is different from the "include" command that includes
contexts within
; other contexts. The #include command works in all asterisk
configuration
files.
;#include "filename.conf"
; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2 ; Trunk interface
;
; Note the 'g2' in the TRUNK variable above. It specifies which group
(defined
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to
use
in
; the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy Zap channel (aka. ascending
sequential hunt group).
; G: select the highest-numbered non-busy Zap channel (aka. descending
sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than
last time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than
last
time (aka. descending rotary hunt group).
;
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]
;
; Any category other than "General" and "Globals" represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal. In pattern
Re: [Asterisk-Users] Why can't sip/200 call sip/202
Sorry to be a pain... but
I restarted my * and now when I launch * get this:
== Parsing '/etc/asterisk/zapata.conf': Found
Jul 24 18:52:45 WARNING[6817]: chan_zap.c:932 zt_open: Unable to specify
channel 1: No such device or address
Jul 24 18:52:45 ERROR[6817]: chan_zap.c:6473 mkintf: Unable to open channel
1: No such device or address
here = 0, tmp->channel = 1, channel = 1
Jul 24 18:52:45 ERROR[6817]: chan_zap.c:10303 setup_zap: Unable to register
channel '1-2'
Jul 24 18:52:45 WARNING[6817]: loader.c:345 ast_load_resource: chan_zap.so:
load_module failed, returning -1
== Unregistered channel type 'Tor'
== Unregistered channel type 'Zap'
Jul 24 18:52:45 WARNING[6817]: loader.c:440 load_modules: Loading module
chan_zap.so failed!
linux:~ # Ouch ... error while writing audio data: : Broken pipe
I have a Junghanns quadBRI card installed. I have modprobe qozap - so it is
loaded and seems to be working OK. I assume there was something in
extensions.conf which was somehow required. something to do with [channels]
?
The error in chan_zap.c seems to be saying that channel 1 cannot be opened.
Angus
- Original Message -
From: "Mark Edwards" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, July 24, 2005 10:13 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
OK Angus
just start here
mv extensions.conf extensions.conf.old
and create a new extensions.conf
[default]
exten => _2XX,1,Dial(SIP/${EXTEN},20,Ttm)
exten => _2XX,2,Hangup
just those 3 lines
do an 'extensions reload' in the CLI or just restart Asterisk
and see if it works
regards,
Mark.
On 7/25/05, Angus Comber <[EMAIL PROTECTED]> wrote:
I think the 777 may be a bit of a Red Herring. I dialed 777 as a test. I
can't dial 202 from 200 if I actually dial 202!
My extensions.conf file:
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
; You can include other config files, use the #include command (without
the
';')
; Note that this is different from the "include" command that includes
contexts within
; other contexts. The #include command works in all asterisk configuration
files.
;#include "filename.conf"
; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2 ; Trunk interface
;
; Note the 'g2' in the TRUNK variable above. It specifies which group
(defined
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to
use
in
; the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy Zap channel (aka. ascending
sequential hunt group).
; G: select the highest-numbered non-busy Zap channel (aka. descending
sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than
last time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than
last
time (aka. descending rotary hunt group).
;
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]
;
; Any category other than "General" and "Globals" represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal. In patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example,
1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
;
; For example the extension _NXX would match normal 7 digit dialings,
; while _1NXXNXX
Re: [Asterisk-Users] Why can't sip/200 call sip/202
You observed correctly. Yes I just copied the sample file, hoping it would
work.
I didn't realise I had to do anything special with the dialplan just for
dialing internal extensions.
Can I use something fairly generic like this (assuming all my extensions are
three digit starting with 2xx):
exten => _2XX,1,Dial(${ARG1})
As a VERY basic first attempt.
By the way can I use (${ARG1}) - is it valid? Or some other variable name
for number dialed?
Is there an Asterisk document on the dialplan. Eg all the variables such as
Dial, Voicemail, etc? Or do we need to look in a certain .h file?
Angus
- Original Message -
From: "dbruce" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, July 24, 2005 10:10 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
The extensions.conf file you provided looks suspiciously like the asterisk
configs/extensions.conf.sample file.
Did you create a dialplan for your specific configuration or did you just
copy the sample file?
----- Original Message -
From: "Angus Comber" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, July 24, 2005 2:50 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
I think the 777 may be a bit of a Red Herring. I dialed 777 as a test.
I
can't dial 202 from 200 if I actually dial 202!
My extensions.conf file:
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
; You can include other config files, use the #include command (without
the
';')
; Note that this is different from the "include" command that includes
contexts within
; other contexts. The #include command works in all asterisk
configuration
files.
;#include "filename.conf"
; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2 ; Trunk interface
;
; Note the 'g2' in the TRUNK variable above. It specifies which group
(defined
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to
use
in
; the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy Zap channel (aka. ascending
sequential hunt group).
; G: select the highest-numbered non-busy Zap channel (aka. descending
sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than
last time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than
last
time (aka. descending rotary hunt group).
;
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]
;
; Any category other than "General" and "Globals" represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal. In patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example,
1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
;
; For example the extension _NXX would match normal 7 digit dialings,
; while _1NXXNXX would represent an area code plus phone number
; preceeded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension.
;
; Contexts contain several lines, one for each step of each
; extension, which can take one of two forms as listed below,
; with the first form being preferred. One may include another
; context in the current one as well, optionally with a
; date and time. I
Re: [Asterisk-Users] Why can't sip/200 call sip/202
Would this do it:
exten => _2XX,1,Dial(${ARG1},30)
Then I would fallback to voicemail (or something else) after the 30 seconds?
Angus
- Original Message -
From: "Marc Storck" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, July 24, 2005 10:06 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
Ok your extensions.conf doesn't mention anything about an extension/number
equal to 202 or 200. You must know that the name of a SIP and IAX2 peer is
only an "address", you will have to assign a number via extensions.conf to
this address.
Have a look at www.voip-info.org and of course google.com to get to know
extensions.conf.
Regards,
Marc
Angus Comber wrote:
I think the 777 may be a bit of a Red Herring. I dialed 777 as a test.
I can't dial 202 from 200 if I actually dial 202!
My extensions.conf file:
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
; You can include other config files, use the #include command (without
the ';')
; Note that this is different from the "include" command that includes
contexts within
; other contexts. The #include command works in all asterisk
configuration files.
;#include "filename.conf"
; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2 ; Trunk interface
;
; Note the 'g2' in the TRUNK variable above. It specifies which group
(defined
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to
use in
; the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy Zap channel (aka. ascending
sequential hunt group).
; G: select the highest-numbered non-busy Zap channel (aka. descending
sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than
last time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than
last time (aka. descending rotary hunt group).
;
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]
;
; Any category other than "General" and "Globals" represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal. In patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example,
1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
;
; For example the extension _NXX would match normal 7 digit dialings,
; while _1NXXNXX would represent an area code plus phone number
; preceeded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension.
;
; Contexts contain several lines, one for each step of each
; extension, which can take one of two forms as listed below,
; with the first form being preferred. One may include another
; context in the current one as well, optionally with a
; date and time. Included contexts are included in the order
; they are listed.
;
;[context]
;exten => someexten,priority,application(arg1,arg2,...)
;exten => someexten,priority,application,arg1|arg2...
;
; Timing list for includes is
;
; |||
;
;include => daytime|9:00-17:00|mon-fri|*|*
;
; ignorepat can be used to instruct drivers to not cancel dialtone upon
; receipt of a particular pattern. The most commonly used example is
; of course '9' like this:
;
;ignorepat => 9
;
; so that dialtone remains even after dialing a 9.
;
;
; Here a
Re: [Asterisk-Users] Why can't sip/200 call sip/202
und(thanks) ; "Thanks for calling press 1 for sales, 2
for support, ..."
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;
;[submenu]
;exten => s,1,Ringing ; Make them comfortable with 2 seconds of ringback
;exten => s,2,Wait,2
;exten => s,3,Background(submenuopts) ; "Thanks for calling the sales
department. Press 1 for steve, 2 for..."
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
include => demo
;
; Extensions like the two below can be used for FWD, Nikotel, sipgate etc.
; Note that you must have a [sipprovider] section in sip.conf whereas
; the otherprovider.net example does not require such a peer definition
;
;exten => _41X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],,r)
;exten => _42X.,1,Dial(SIP/user:[EMAIL PROTECTED]:[EMAIL PROTECTED],30,rT)
; Real extensions would go here. Generally you want real extensions to be 4
or 5
; digits long (although there is no such requirement) and start with a
single
; digit that is fairly large (like 6 or 7) so that you have plenty of room
to
; overlap extensions and menu options without conflict. You can alias them
with
; names, too and use global variables
;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1 ; Channel hints for presence
;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
;exten => 6245,1,Dial(${HINT},20,rtT) ; Use hint as listed
;exten => 6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time limit
;exten => 6389,1,Dial(MGCP/aaln/[EMAIL PROTECTED])
;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK}
;exten => 6275,1,Macro(stdexten,6275,${MARK}) ; assuming ${MARK} is
something like Zap/2
;exten => mark,1,Goto(6275|1) ; alias mark to 6275
;exten => 6536,1,Macro(stdexten,6236,${WIL}) ; Ditto for wil
;exten => wil,1,Goto(6236|1)
;
; Some other handy things are an extension for checking voicemail via
; voicemailmain
;
;exten => 8500,1,VoicemailMain
;exten => 8500,2,Hangup
;
; Or a conference room (you'll need to edit meetme.conf to enable this room)
;
;exten => 8600,1,Meetme(1234)
;
; Or playing an announcement to the called party, as soon it answers
;
;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
;
; For more information on applications, just type "show applications" at
your
; friendly Asterisk CLI prompt.
;
; 'show application ' will show details of how you
; use that particular application in this file, the dial plan.
;
- Original Message -
From: "dbruce" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, July 24, 2005 8:39 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
Marc: My answer is not incorrect... it is incomplete.
The OP stipulated 2 extensions 200 and 202... and provided a sip debug
indicating a call from 200 to 777.
I pointed out the obvious.
If the OP is dialing 202 on the phone, and the phone is dialing 777, then
he
needs to look at the dialplan configuration of the phone. If he is dialing
777 on the phone and expecting to reach 202, then he will need to have
translations in the asterisk dialplan. But, the question was "what should
I
be looking at?"... Using just the information provided, and the fact that
he
is new to asterisk... without any further information... the first thing
he
should be looking at is why the phone is trying to reach 777 when he wants
to reach 202... Many new users do not realize the complexity of the SIP
protocol, and only really look at the trace in a general manner... such
as:
INVITE
407 Proxy Authentication Required
ACK
INVITE
404 Not Found
ACK
The idea was to provide a clue... not to provide a complete working
dialplan
and phone configuration. Providing new users with "the complete package"
is
a dis-service to them. They will only learn from thier mistakes and
experiences.. providing clues allows them to expand their experience and
build their confidence... It requires them to look at the details and
learn
to analyse them.
Regards,
Derek
- Original Message -
From: "Marc Storck" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, July 24, 2005 12:53 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
Derek: you reply is uncorrect. If Angus has the extension 777 in his
dialplan/extensions.conf which will dial 202. The name of the peer has
absolutely nothing to do with which number/name he would have to dial.
Without dialplan he will be unable to call any extension even 202, as
202 is only the name of the peer.
Angus: please paste your extensions.conf to pastebin.ca
Regards,
Marc
dbruce wrote:
> It appears from the d
[Asterisk-Users] Why can't sip/200 call sip/202
I have 2 sip accounts setup - 200 and 202. If I do sip show peers I get: sip show peersName/username Host Dyn Nat ACL Mask Port Status202/202 192.168.0.6 D 255.255.255.255 5060 Unmonitored201/201 (Unspecified) D 255.255.255.255 5060 Unmonitored200/200 192.168.0.3 D 255.255.255.255 5060 Unmonitored 200 is a Grandstream GXP200 IP Phone and 202 is a Grandstream BT100 IP phone. relevant bit of sip.conf: [200]username=200type=friendsecret=1234port=5060nat=neverdtmfmode=rfc2833context=defaultcallerid="Angus Comber" <200>host=dynamicdisallow=allallow=ulawallow=alawallow=g723.1allow=g729 [202]username=202type=friendsecret=1234port=5060nat=neverdtmfmode=rfc2833context=defaultcallerid="Sam Comber" <202>host=dynamicdisallow=allallow=ulawallow=alawallow=g723.1allow=g729 But whenever I try to dial between phones I get this: Sip read: 0 headers, 0 lines Sip read:INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1From: "Angus Comber" ;tag=a1afaf4fdb0ac845To: Contact: Supported: replaces, timerCall-ID: [EMAIL PROTECTED]CSeq: 45925 INVITEUser-Agent: Grandstream GXP2000 1.0.1.9Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACKContent-Type: application/sdpContent-Length: 258 v=0o=200 8000 8000 IN IP4 192.168.0.3s=SIP Callc=IN IP4 192.168.0.3t=0 0m=audio 5004 RTP/AVP 18 0 8 101a=sendrecva=rtpmap:18 G729/8000a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-11 13 headers, 13 linesUsing latest request as basis requestSending to 192.168.0.3 : 5060 (non-NAT)Reliably Transmitting (no NAT):SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1From: "Angus Comber" ;tag=a1afaf4fdb0ac845To: ;tag=as668982beCall-ID: [EMAIL PROTECTED]CSeq: 45925 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: Proxy-Authenticate: Digest realm="asterisk", nonce="0c555366"Content-Length: 0 to 192.168.0.3:5060Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 msFound user '200' Sip read:ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1From: "Angus Comber" ;tag=a1afaf4fdb0ac845To: ;tag=as668982beContact: Call-ID: [EMAIL PROTECTED]CSeq: 45925 ACKUser-Agent: Grandstream GXP2000 1.0.1.9Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACKContent-Length: 0 11 headers, 0 lines Sip read:INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304From: "Angus Comber" ;tag=a1afaf4fdb0ac845To: Contact: Supported: replaces, timerProxy-Authorization: Digest username="200", realm="asterisk", algorithm=MD5, uri="sip:[EMAIL PROTECTED];user=phone", nonce="0c555366", response="ee6088fb4e50da5fe412913ae40dd45c"Call-ID: [EMAIL PROTECTED]CSeq: 45926 INVITEUser-Agent: Grandstream GXP2000 1.0.1.9Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACKContent-Type: application/sdpContent-Length: 258 v=0o=200 8000 8001 IN IP4 192.168.0.3s=SIP Callc=IN IP4 192.168.0.3t=0 0m=audio 5004 RTP/AVP 18 0 8 101a=sendrecva=rtpmap:18 G729/8000a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=ptime:20a=rtpmap:101 telephone-event/8000a=fmtp:101 0-11 14 headers, 13 linesUsing latest request as basis requestSending to 192.168.0.3 : 5060 (non-NAT)Found user '200'Found RTP audio format 18Found RTP audio format 0Found RTP audio format 8Found RTP audio format 101Peer audio RTP is at port 192.168.0.3:5004Found description format G729Found description format PCMUFound description format PCMAFound description format telephone-eventCapabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)Looking for 777 in defaultReliably Transmitting (no NAT):SIP/2.0 404 Not FoundVia: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304From: "Angus Comber" ;tag=a1afaf4fdb0ac845To: ;tag=as668982beCall-ID: [EMAIL PROTECTED]CSeq: 45926 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: Content-Length: 0 to 192.168.0.3:5060 Sip read:ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304From: "Angus Comber" ;tag=a1afaf4fdb0ac845To: ;tag=as668982beContact: Proxy-Authorization: Digest username="200", realm="asterisk", algorithm
[Asterisk-Users] Do I have to worry about interrupt sharing here?
Hello I am using a Junghanns QuadBRI ISDN card - the module name is qozap. If I like at my interrupt assignment, qozap is sharing interrupt 10 with libata and uhci_hcd. I think libata is the IDE hard drive module and uhci_hcd is a USB module. linux:~ # modprobe qozaplinux:~ # cat /proc/interrupts CPU0 0: 12634579 XT-PIC timer 1: 10 XT-PIC i8042 2: 0 XT-PIC cascade 3: 0 XT-PIC Intel ICH5 5: 0 XT-PIC uhci_hcd 7: 0 XT-PIC parport0 8: 2 XT-PIC rtc 9: 21988 XT-PIC acpi, ehci_hcd, eth0 10: 7657 XT-PIC libata, uhci_hcd, qozap 11: 0 XT-PIC uhci_hcd, uhci_hcd 12: 118 XT-PIC i8042 14: 54851 XT-PIC ide0 15: 25272 XT-PIC ide1NMI: 0LOC: 0ERR: 0MIS: 0 Should I disable USB in the BIOS? Should that remove uhci_hcd loading? Is there a way to re-allocate the interrupt used for libata? Angus ___ 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
Re: [Asterisk-Users] Problems installing asterisk-addons
How strange - that worked! I wonder why that was put there? Angus - Original Message - From: "Tzafrir Cohen" <[EMAIL PROTECTED]> To: Sent: Friday, July 22, 2005 8:29 AM Subject: Re: [Asterisk-Users] Problems installing asterisk-addons On Thu, Jul 21, 2005 at 09:45:02PM +0100, Angus Comber wrote: I am now getting this make error: cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory Remove the line that includes asterisk.h . Doesn't help anybody. This is basically the patch I needed to apply to asterisk-addons to make it build with the debian package asterisk-devel . -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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
Re: [Asterisk-Users] Problems installing asterisk-addons
I am now getting this make error: cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory make: *** [cdr_addon_mysql.o] Error 1 linux:/usr/src/asterisk-addons # But I have the asterisk sources in /usr/include/asterisk but I am installing asterisk-addons from /usr/src/asterisk-addons/ Is that a problem? I think the problem is in line 29 - #include "asterisk.h" of cdr_addon_mysql.c . I assume that I should not really have to edit any of the source or make files. I bet something fairly basic is wrong. any ideas? Angus - Original Message - From: "Tzafrir Cohen" <[EMAIL PROTECTED]> To: Sent: Thursday, July 21, 2005 2:55 PM Subject: Re: [Asterisk-Users] Problems installing asterisk-addons On Thu, Jul 21, 2005 at 03:44:03PM +0200, Christoph Eicke wrote: On Thursday 21 July 2005 15:28, Angus Comber wrote: > My asterisk version is Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j > > It is a version put together by Junghanns.net - for working with their > ISDN > cards. Mmm I wonder if that is the problem? If so then what version > of > asterisk-addons do I install. I didn't see anything about > asterisk-addons > on the junghanns.net site. You are right, that is the problem. I wasn't able to compile the addons with the version from junghanns.net. I suspect that it's because those addons compile the MySQL realtime extension and the Asterisk version coming with the bristuff package has no support for the realtime extension yet. 1.0.9 has no support for realtime yet, both in addon in in the main distribution. You seem to be mixing 1.0 and HEAD. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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
Re: [Asterisk-Users] Problems installing asterisk-addons
I was installing 1.0.9 asterisk-addons on a Suse Professional 9.3 installation. Angus - Original Message - From: Mohamed A. Gombolaty To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, July 21, 2005 12:39 PM Subject: Re: [Asterisk-Users] Problems installing asterisk-addons Hi Angus, I don't believe it can be the root password of mysql, I used to install the addons without even haved installed mysql server yet, I guess we need to know which platform are you working on and which version you are trying to install. Thx MAG Angus Comber wrote: Hello I have downloaded asterisk-addons but when I make install get: cc -fPIC -I../asterisk -D_GNU_SOURCE -DMYSQL_LOGUNIQUEID -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 I have set a password for root on mysql - could that be the problem? Should I remove the password? What is easiest way to do that? Angus ___ 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-- Thx MAG ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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
Re: [Asterisk-Users] Problems installing asterisk-addons
My asterisk version is Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j It is a version put together by Junghanns.net - for working with their ISDN cards. Mmm I wonder if that is the problem? If so then what version of asterisk-addons do I install. I didn't see anything about asterisk-addons on the junghanns.net site. Angus - Original Message - From: "Dave Cotton" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, July 21, 2005 12:56 PM Subject: Re: [Asterisk-Users] Problems installing asterisk-addons On Thu, 2005-07-21 at 12:19 +0100, Angus Comber wrote: Hello I have downloaded asterisk-addons but when I make install get: cc -fPIC -I../asterisk -D_GNU_SOURCE -DMYSQL_LOGUNIQUEID -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 I have set a password for root on mysql - could that be the problem? Should I remove the password? What is easiest way to do that? You haven't got far enough for that to be a problem, that would be at runtime. Are your asterisk and asterisk-addons in sync? i.e. the same release, you're not trying to mix HEAD and stable are you? -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems installing asterisk-addons
Hello I have downloaded asterisk-addons but when I make install get: cc -fPIC -I../asterisk -D_GNU_SOURCE -DMYSQL_LOGUNIQUEID -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.capp_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 givenapp_addon_sql_mysql.c: In function `del_identifier':app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use in this function)app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported only onceapp_addon_sql_mysql.c:164: error: for each function it appears in.)make: *** [app_addon_sql_mysql.o] Error 1 I have set a password for root on mysql - could that be the problem? Should I remove the password? What is easiest way to do that? Angus ___ 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
Re: [Asterisk-Users] Mahler's Book - New Project
Personally I wouldn't bother with the Mahler book. I bought it in the hope that it might be the panacea I was looking for. It wasn't. If you read it you will recognise a lot of the standard text you will see on Digium or other web sites. If I had time I would write the book myself. I did find a useful handbook - but can't find it right now - or where I found it. Asterisk @ Home is a quick way to get up and running first. But to really get to know the product you are best to have a go doing it the hard way - ie installing a brand of Linux and getting familiar with the conf files. Having a strong linux knowledge helps a lot. Best way to learn as always is to buy a telephony card and set up for real. Angus - Original Message - From: David Stude To: asterisk-users@lists.digium.com Sent: Wednesday, July 20, 2005 2:56 PM Subject: [Asterisk-Users] Mahler's Book - New Project Hi all, I'm currently gearing up for a possible PBX replacement project using Asterisk, and I'm just breaching the iceberg of information that's available. I typically like to have something thick with pages in front of me. Mahler's book was the first one to come up and it seems like a good place to start. However, the big name bookstores tell me it'll take up to three weeks, and this project simply can't endure that wait. Does anyone know where it's possible to get a paper copy *quickly*? #2, I'm planning to interface Asterisk with a Norstar MICS via PRI. Can anyone recommend a reference book or site more suited to this task? Thanks and regards, David Stude Receptec, LLC Holly, MI ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and flash disks
Hello I see it is possible to buy Flash Disks up to 4GB now. Has anyone any experience of building an Asterisk system with a flash disk as the only storage device? Any brands you recommend? Is 2 or 4GB enough for an Asterisk installation? Typically how many MB is required for voicemail recording files for say a 10 user system? What about voicemail - I suppose files could be emailed and deleted immediately? Angus Comber [EMAIL PROTECTED] ___ 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] channel.c:41:31: asterisk/transcap.h: No such file or directory problem
Hello I am trying to get Asterisk to work with the Junghanns Quad BRI ISDN card. I am progressing slowly! Problem I am now experiencing is as below. gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"1.0.8-BRIstuffed-0.2.0-RC8h\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN -c -o channel.o channel.cchannel.c:41:31: asterisk/transcap.h: No such file or directorychannel.c: In function `ast_transfercapability2str':channel.c:239: error: `AST_TRANS_CAP_SPEECH' undeclared (first use in this function)channel.c:239: error: (Each undeclared identifier is reported only oncechannel.c:239: error: for each function it appears in.)channel.c:241: error: `AST_TRANS_CAP_DIGITAL' undeclared (first use in this function)channel.c:243: error: `AST_TRANS_CAP_RESTRICTED_DIGITAL' undeclared (first use in this function)channel.c:245: error: `AST_TRANS_CAP_3_1K_AUDIO' undeclared (first use in this function)channel.c:247: error: `AST_TRANS_CAP_DIGITAL_W_TONES' undeclared (first use in this function)channel.c:249: error: `AST_TRANS_CAP_VIDEO' undeclared (first use in this function)channel.c: In function `ast_channel_bridge':channel.c:2623: warning: implicit declaration of function `IS_DIGITAL'make: *** [channel.o] Error 1 ASTERISK installed. Installation finished.Is the problem here the line: channel.c:41:31: asterisk/transcap.h: No such file or directory ?? Do I just need to ger hold of transcap.h? Or something else?Angus Comber[EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] Dial *97 to pickup voicemail buts saysmypasswordincorrect
I had two SIP phones, it worked on one and not the other so that helped. I had to set Send DTMF: to be via RTP (RFC2833) (not in-audio). Angus - Original Message - From: "Chris Coulthurst" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Monday, July 04, 2005 3:56 PM Subject: RE: [Asterisk-Users] Dial *97 to pickup voicemail buts saysmypasswordincorrect Not sure why I see *97 and *98 here, but I would check your dtmfmode= line in sip.conf. Often times, using rfc2833 works when inband or sip-info doesn't. See http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+dtmfmode Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Angus Comber |Sent: Monday, July 04, 2005 4:34 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Dial *97 to pickup voicemail |buts says mypasswordincorrect | | |I have found that if I dial from another extension *98 and |select extn 200 |and enter password 1234 it works. So is it something to do with |configuration on my IP Phone? It is a Grandstream GXP2000 running: |Software Version: Program-- 1.0.0.3Bootloader-- 1.0.0.3 | |Anyone got any ideas? | |Angus | | | |----- Original Message - |From: Angus Comber |To: asterisk-users@lists.digium.com |Sent: Monday, July 04, 2005 12:20 PM |Subject: [Asterisk-Users] Dial *97 to pickup voicemail buts says my |passwordincorrect | | |Hello | |I am at extension 200 and I know there is a voicemail message |waiting. I |dial *97 and am prompted for the password. I enter 1234 which |I have set as |my voicemail password. What can I do to troubleshoot? | |Angus Comber |Itel Office Software Ltd |5 Enmore Gardens |London, SW14 8RF |Tel: 020 8878 7367 |Fax: 020 8876 7257 |Em: [EMAIL PROTECTED] |web: www.iteloffice.com | | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-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/asteri|sk-users |To |UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial *97 to pickup voicemail buts says my passwordincorrect
I have found that if I dial from another extension *98 and select extn 200 and enter password 1234 it works. So is it something to do with configuration on my IP Phone? It is a Grandstream GXP2000 running: Software Version: Program-- 1.0.0.3Bootloader-- 1.0.0.3 Anyone got any ideas? Angus - Original Message - From: Angus Comber To: asterisk-users@lists.digium.com Sent: Monday, July 04, 2005 12:20 PM Subject: [Asterisk-Users] Dial *97 to pickup voicemail buts says my passwordincorrect Hello I am at extension 200 and I know there is a voicemail message waiting. I dial *97 and am prompted for the password. I enter 1234 which I have set as my voicemail password. What can I do to troubleshoot? Angus Comber Itel Office Software Ltd 5 Enmore Gardens London, SW14 8RF Tel: 020 8878 7367 Fax: 020 8876 7257 Em: [EMAIL PROTECTED] web: www.iteloffice.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial *97 to pickup voicemail buts says my password incorrect
Hello I am at extension 200 and I know there is a voicemail message waiting. I dial *97 and am prompted for the password. I enter 1234 which I have set as my voicemail password. What can I do to troubleshoot? Angus ComberItel Office Software Ltd5 Enmore GardensLondon, SW14 8RFTel: 020 8878 7367Fax: 020 8876 7257Em: [EMAIL PROTECTED]web: www.iteloffice.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can I dial a number from handset to pickup voicemail?
Hello Maybe a silly question, but after some searching couldn't find answer. Is there a number I can dial to pickup and listen to my voicemail messages on my SIP phone? I am used to eg dialling *17 to pickup my voicemail messages on Avaya system? Angus ___ 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
Re: [Asterisk-Users] How can you check that egTDM04B hardwareinstalled and drivers OK
Digium logged in and fixed the problem. It seems they had to fix the zaptel source code - so not really something I could easily have done. something about adding the subvendors ID to the cards source. So I assume a bug. I personally feel a little indebted to Digium for sorting the problem and obviously making the Asterisk available. But would like them even more if I didn't have to go through these problems ;) Angus - Original Message - From: "John Novack" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, June 22, 2005 2:08 PM Subject: Re: [Asterisk-Users] How can you check that egTDM04B hardwareinstalled and drivers OK Probably means that your perfectly good motherboard can't see the TDM card. There are many motherboards that this card doesn't seem to work with, Digium doesn't seem willing to address the issue or even acknowledge that is the case, and usually answers " try another motherboard" rather than 'fess up that there is a design problem with the PCI interface and correct it. PCI 2.2 is a stated requirement, but there is certainly more to the story than that. In addition, when the board CAN be seen, report rev E/F when the silkscreen reads Rev H, someone mentioned there is now a Rev I ( good luck getting an exchange ) and Digium 's answer is " if we can see it through remote access" then there is no reason to replace it, and if we can't, try another MB. Overall, if it works, lucky you, if not, Too bad. Hard to support Digium and suggest others purchase such a product. Best you look for other interfaces to Asterisk. John Novack Angus Comber wrote: If I try dmesg - no mention of a Wildcard TDM400. Sorry I am fairly new to Linux. In Windows I suppose I would run some hardware program which came with the card to see if I could manually set IRQ's etc. What should I be looking at now? Please feel free to point me to a good book or whatever you feel is appropriate. Could the card be faulty? My motherboard is an Intel D865GLC. I am running [EMAIL PROTECTED] version 1.0 Angus - Original Message - From: "Mike M" <[EMAIL PROTECTED]> To: Sent: Tuesday, June 21, 2005 3:14 PM Subject: Re: [Asterisk-Users] How can you check that eg TDM04B hardwareinstalled and drivers OK On Tue, Jun 21, 2005 at 07:09:46AM -0600, Rich Adamson wrote: > I am struggling to get my TDM04B working. Just to rule out a hardware > problem how can I check that the hardware works? How can I then > check that the drivers are loaded correctly? > 1. from the linux command line, type 'dmesg' and look for Found a Wildcard TDM: Wildcard TDM400P REV H (4 modules) if you see that, the TDM card is recognized by the OS. Here's what I get on a working system: [EMAIL PROTECTED] src]# modprobe wctdm [EMAIL PROTECTED] src]# Jun 21 10:10:55 b2 kernel: PCI: Found IRQ 11 for device 01:0a.0 Jun 21 10:10:55 b2 kernel: Freshmaker version: 71 Jun 21 10:10:55 b2 kernel: Freshmaker passed register test Jun 21 10:10:55 b2 kernel: Module 0: Installed -- AUTO FXS/DPO Jun 21 10:10:55 b2 kernel: Module 1: Not installed Jun 21 10:10:55 b2 kernel: Module 2: Not installed Jun 21 10:10:55 b2 kernel: Module 3: Not installed Jun 21 10:10:55 b2 kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Jun 21 10:10:55 b2 kernel: Registered tone zone 0 (United States / North America) [EMAIL PROTECTED] src]# cat /proc/interrupts CPU0 0: 17893766 XT-PIC timer 1: 2 XT-PIC keyboard 2: 0 XT-PIC cascade 4: 357411641 XT-PIC eth0, wanpipe1 8: 1 XT-PIC rtc 10:3381408 XT-PIC Intel ICH2 11: 178236906 XT-PIC wctdm 14: 50492 XT-PIC ide0 15: 0 XT-PIC ide1 NMI: 0 ERR: 0 [EMAIL PROTECTED] src]# cat /proc/interrupts CPU0 0: 17894203 XT-PIC timer 1: 2 XT-PIC keyboard 2: 0 XT-PIC cascade 4: 357419974 XT-PIC eth0, wanpipe1 8: 1 XT-PIC rtc 10:3381408 XT-PIC Intel ICH2 11: 178241275 XT-PIC wctdm 14: 50494 XT-PIC ide0 15: 0 XT-PIC ide1 NMI: 0 ERR: 0 -- Mike ___ 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
Re: [Asterisk-Users] How can you check that eg TDM04B hardwareinstalled and drivers OK
Hello Here is what I find. Any help would be greatly appreciated. Angus - Original Message - From: "Rich Adamson" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, June 21, 2005 2:09 PM Subject: Re: [Asterisk-Users] How can you check that eg TDM04B hardwareinstalled and drivers OK I am struggling to get my TDM04B working. Just to rule out a hardware problem how can I check that the hardware works? How can I then check that the drivers are loaded correctly? You didn't mention which linux distro you're using, so translate the ** [EMAIL PROTECTED] version 1.0 on Centos OS. following into whatever your system expects. Try the following items: 1. from the linux command line, type 'dmesg' and look for Found a Wildcard TDM: Wildcard TDM400P REV H (4 modules) if you see that, the TDM card is recognized by the OS. ** Did not find TDM! 2. from the linux command line, type 'cat /proc/interrupts and look for an entry with 'wctdm' in the list. If you don't see wctdm listed, the module is not loaded as yet. ** no wctdm in list 3. in /etc/zaptel.conf, ensure you have an entry like: fxsks=1-4 ** OK - but think a hardware issue needs to be resolved first 4. if you're using a linux v2.6 kernel, read /usr/src/zaptel/README.udev 5. with asterisk stopped and from the linux command line, try sysconfig zaptel start ** Command not found 6. What do you see if you run 'zttool' from the linux command line? ** ** Zaptel Tool loads and I see this: Zapata Telephony Interfaces Alarms Span nothing else If click on Select go to another screen: Current Alarms: No Alarms Sync Source: Internally clocked IRQ Misses: 0 Bipolar Viol: 0 Tx/Rx Levels: 0/ 0 Total/Conf/Act: 0/ 0/ 0 ___ 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
Re: [Asterisk-Users] How can you check that eg TDM04B hardwareinstalled and drivers OK
If I try dmesg - no mention of a Wildcard TDM400. Sorry I am fairly new to Linux. In Windows I suppose I would run some hardware program which came with the card to see if I could manually set IRQ's etc. What should I be looking at now? Please feel free to point me to a good book or whatever you feel is appropriate. Could the card be faulty? My motherboard is an Intel D865GLC. I am running [EMAIL PROTECTED] version 1.0 Angus - Original Message - From: "Mike M" <[EMAIL PROTECTED]> To: Sent: Tuesday, June 21, 2005 3:14 PM Subject: Re: [Asterisk-Users] How can you check that eg TDM04B hardwareinstalled and drivers OK On Tue, Jun 21, 2005 at 07:09:46AM -0600, Rich Adamson wrote: > I am struggling to get my TDM04B working. Just to rule out a hardware > problem how can I check that the hardware works? How can I then > check that the drivers are loaded correctly? > 1. from the linux command line, type 'dmesg' and look for Found a Wildcard TDM: Wildcard TDM400P REV H (4 modules) if you see that, the TDM card is recognized by the OS. Here's what I get on a working system: [EMAIL PROTECTED] src]# modprobe wctdm [EMAIL PROTECTED] src]# Jun 21 10:10:55 b2 kernel: PCI: Found IRQ 11 for device 01:0a.0 Jun 21 10:10:55 b2 kernel: Freshmaker version: 71 Jun 21 10:10:55 b2 kernel: Freshmaker passed register test Jun 21 10:10:55 b2 kernel: Module 0: Installed -- AUTO FXS/DPO Jun 21 10:10:55 b2 kernel: Module 1: Not installed Jun 21 10:10:55 b2 kernel: Module 2: Not installed Jun 21 10:10:55 b2 kernel: Module 3: Not installed Jun 21 10:10:55 b2 kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Jun 21 10:10:55 b2 kernel: Registered tone zone 0 (United States / North America) [EMAIL PROTECTED] src]# cat /proc/interrupts CPU0 0: 17893766 XT-PIC timer 1: 2 XT-PIC keyboard 2: 0 XT-PIC cascade 4: 357411641 XT-PIC eth0, wanpipe1 8: 1 XT-PIC rtc 10:3381408 XT-PIC Intel ICH2 11: 178236906 XT-PIC wctdm 14: 50492 XT-PIC ide0 15: 0 XT-PIC ide1 NMI: 0 ERR: 0 [EMAIL PROTECTED] src]# cat /proc/interrupts CPU0 0: 17894203 XT-PIC timer 1: 2 XT-PIC keyboard 2: 0 XT-PIC cascade 4: 357419974 XT-PIC eth0, wanpipe1 8: 1 XT-PIC rtc 10:3381408 XT-PIC Intel ICH2 11: 178241275 XT-PIC wctdm 14: 50494 XT-PIC ide0 15: 0 XT-PIC ide1 NMI: 0 ERR: 0 -- Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How can you check that eg TDM04B hardware installed and drivers OK
Hello I am struggling to get my TDM04B working. Just to rule out a hardware problem how can I check that the hardware works? How can I then check that the drivers are loaded correctly? Angus ___ 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] Can't get TDM04B to work!
Can't get a Digium TDM04B working. Asterisk is running. I seem to have setup the trunks OK. But whenever I make an outgoing call get the 'all circuits are busy now' message. If I call in nothing happens at all! Here is my zapata.conf file: ;; Zapata telephony interface;; Configuration file [trunkgroups] [channels] language=encontext=from-pstnsignalling=fxs_ksfxsks=1-4rxwink=300 ; Atlas seems to use long (250ms) winks;; Whether or not to do distinctive ring detection on FXO lines;;usedistinctiveringdetection=yes usecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=noechotraining=800rxgain=0.0txgain=0.0group=0callgroup=1pickupgroup=1immediate=no ;faxdetect=bothfaxdetect=incoming;faxdetect=outgoing;faxdetect=no ;Include AMP configs#include zapata_additional.conf ;Include genzaptelconf configs#include zapata-auto.conf What am I doing wrong? Angus Comber [EMAIL PROTECTED] ___ 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] Do I need a ring capacitor to use TDM400P cards in UK
Hello I have played about with a TDM400 card and plugged in some standard analog phones. I am using the card in FXS mode - for analog extensions. I did notice that one of my phones did not ring and I wondered why. I later read in Paul Mahler's book VoIP Telephony with Asterisk that in his section on the TDM400 on page 127 he says "In the UK, you may need an adapter that provides a ring capacitor, or the phone may not ring." Can anyone confirm this. Also what is one of those and where would I find a good supplier? I am in the trade so wholesale would be OK. Angus Comber ___ 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] ISDN 4 BRI card for UK
Hello I want to setup an Asterisk in several offices with 4 BRI ISDN. I am looking for recommendations on hardware. Criteria would be ease of setup, reliability and cost. The Eicon 4 BRI cards seem fairly pricey. Shame Digium don't do a ISDN BRI card. Angus ___ 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
Re: [Asterisk-Users] X100P installed OK, after added TDM400P Asterisk would no longer start
- Original Message - From: "Ralf Schlatterbeck" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Sunday, June 05, 2005 6:22 AM Subject: Re: [Asterisk-Users] X100P installed OK,after added TDM400P Asterisk would no longer start On Sat, Jun 04, 2005 at 11:20:47PM +0100, Angus Comber wrote: This is what I have: # Span 1: WCTDM/0 "Wildcard TDM400P REV H Board 1" fxoks=1 fxoks=2 fxoks=3 # channel 4, WCTDM, inactive. # Span 2: WCFXO/0 "Wildcard X101P Board 1" fxsks=5 # Global data loadzone = us defaultzone = us Which from your response appears correct. Could it have confused the cards? If I swap them in the config might that work? Depends on the order you load the drivers I think. So swapping them in the config will probably help, yes. The FXS modules would then be 5-7 and the FXO 1... You really have two boards, one with 3 FXS modules and one with a single FXO module? In the end I just removed the X100P board. I want to buy a four port analog trunk card and use this for real anyway. I suspect however that if I had swapped the cards in the config it might have worked. Angus Ralf -- Ralf Schlatterbeck email: [EMAIL PROTECTED] FAX: +43/2243/26465/23 ___ 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
Re: [Asterisk-Users] X100P installed OK, after added TDM400P Asterisk would no longer start
This is what I have: # Span 1: WCTDM/0 "Wildcard TDM400P REV H Board 1" fxoks=1 fxoks=2 fxoks=3 # channel 4, WCTDM, inactive. # Span 2: WCFXO/0 "Wildcard X101P Board 1" fxsks=5 # Global data loadzone = us defaultzone = us Which from your response appears correct. Could it have confused the cards? If I swap them in the config might that work? Angus - Original Message - From: "Ralf Schlatterbeck" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, June 04, 2005 8:30 PM Subject: Re: [Asterisk-Users] X100P installed OK,after added TDM400P Asterisk would no longer start On Sat, Jun 04, 2005 at 07:57:36PM +0100, Angus Comber wrote: But I have just added a TDM400P card (specifically a TDM30B) and now problems. Found a Wildcard TDM: Wildcard TDM400 R Rev H (4 module) wcfxs Running ztcfg: ZT_CHANCONFIG failed on channel 1: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling [FAILED] You have FXS modules (to attach analogue phones), the driver already tells you, you probably have configured wrong signalling in /etc/zaptel.conf, for an FXS module it should have somthing like: fxoks=1-8 (the numbers depend on how many modules you have, so you should have 1-3 I guess) Ralf -- Ralf Schlatterbeck email: [EMAIL PROTECTED] FAX: +43/2243/26465/23 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P installed OK, after added TDM400P Asterisk would no longer start
Hello I setup [EMAIL PROTECTED] with purely VoIP and it worked fine. I then added an X100P card so I could call out / take inbound calls via PSTN and that went fine. But I have just added a TDM400P card (specifically a TDM30B) and now problems. Here is some of the output. Any ideas on what I should be looking at next? When I run genzaptelconf -s -d I get lots of erors on screen - bit I can see now is: Found a Wildcard TDM: Wildcard TDM400 R Rev H (4 module) wcfxs Running ztcfg: ZT_CHANCONFIG failed on channel 1: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling [FAILED] STARTING ASTERISK Asterisk ended with exit status 1 Asterisk died with code 1 Automatically restarting Asterisk Asterisk ended with exit status 1 Asteirsk died with code 1 Automatically restarting Asterisk - Asterisk could not start! use tail etc Output was: [EMAIL PROTECTED] root]# tail /var/log/asterisk/full Jun 4 14:35:34 VERBOSE[2223]: == Parsing '/etc/asterisk/zapata_additional.con f': Jun 4 14:35:34 VERBOSE[2223]: == Parsing '/etc/asterisk/zapata_additional.conf': Found Jun 4 14:35:34 VERBOSE[2223]: == Parsing '/etc/asterisk/zapata-auto.conf': Jun 4 14:35:34 VERBOSE[2223]: == Parsing '/etc/asterisk/zapata-auto.conf': Found Jun 4 14:35:34 WARNING[2223]: Unable to specify channel 1: No such device or address Jun 4 14:35:34 ERROR[2223]: Unable to open channel 1: No such device or address here = 0, tmp->channel = 1, channel = 1 Jun 4 14:35:34 ERROR[2223]: Unable to register channel '1' Jun 4 14:35:34 WARNING[2223]: chan_zap.so: load_module failed, returning -1 Jun 4 14:35:34 VERBOSE[2223]: == Unregistered channel type 'Tor' Jun 4 14:35:34 VERBOSE[2223]: == Unregistered channel type 'Zap' Jun 4 14:35:34 WARNING[2223]: Loading module chan_zap.so failed! [EMAIL PROTECTED] root]# [EMAIL PROTECTED] root]# tail /var/log/asterisk/full Jun 4 14:35:34 WARNING[2223]: chan_zap.so: load_module failed, returning -1 Jun 4 14:35:34 VERBOSE[2223]: == Unregistered channel type 'Tor' Jun 4 14:35:34 VERBOSE[2223]: == Unregistered channel type 'Zap' -bash: [EMAIL PROTECTED]: command not found Jun 4 14:35:34 WARNING[2223]: Loading module chan_zap.so failed! [EMAIL PROTECTED] root]# Jun 4 14:35:34 VERBOSE[2223]: == Parsing '/etc/asterisk/zapata_additional.con f': Jun 4 14:35:34 VERBOSE[2223]: == Parsing '/etc/asterisk/zapata_additional .conf': Found -bash: Jun: command not found [EMAIL PROTECTED] root]# Jun 4 14:35:34 VERBOSE[2223]: == Parsing '/etc/asterisk/zapata-auto.conf': Ju -bash: Jun: command not found [EMAIL PROTECTED] root]# n 4 14:35:34 VERBOSE[2223]: == Parsing '/etc/asterisk/zapata-auto.conf': Foun -bash: n: command not found [EMAIL PROTECTED] root]# d -bash: d: command not found [EMAIL PROTECTED] root]# Jun 4 14:35:34 WARNING[2223]: Unable to specify channel 1: No such device or ad -bash: Jun: command not found [EMAIL PROTECTED] root]# dress -bash: dress: command not found [EMAIL PROTECTED] root]# Jun 4 14:35:34 ERROR[2223]: Unable to open channel 1: No such device or address -bash: Jun: command not found [EMAIL PROTECTED] root]# here = 0, tmp->channel = 1, channel = 1 -bash: here: command not found [EMAIL PROTECTED] root]# Jun 4 14:35:34 ERROR[2223]: Unable to register channel '1' -bash: Jun: command not found [EMAIL PROTECTED] root]# Jun 4 14:35:34 WARNING[2223]: chan_zap.so: load_module failed, returning -1 -bash: Jun: command not found [EMAIL PROTECTED] root]# Jun 4 14:35:34 VERBOSE[2223]: == Unregistered channel type 'Tor' -bash: Jun: command not found [EMAIL PROTECTED] root]# Jun 4 14:35:34 VERBOSE[2223]: == Unregistered channel type 'Zap' -bash: Jun: command not found [EMAIL PROTECTED] root]# Jun 4 14:35:34 WARNING[2223]: Loading module chan_zap.so failed! -bash: Jun: command not found [EMAIL PROTECTED] root]# ___ 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] Is such a thing as a analog (or even IP) video door entry system available?
I want to setup a video door entry system. I understand a lot of the systems on the market use proprietary technology. But ideally if the system could connect into a normal analog port or even use IP to my Asteirsk that would be a lot better. Then I could have video phones on users desks so anyone can see who is at the door. Anyone aware of any suitable products. Angus Comber [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] faxes
How does a Windows workstation fax via Asterisk? Has someone written a Asterisk fax print driver? Or some other way? Angus Comber [EMAIL PROTECTED] - Original Message - From: "Henry Devito" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, March 26, 2005 9:11 AM Subject: Re: [Asterisk-Users] faxes I've been working on, actually just started, creating a network app where windoze pc's can print to a virtual printer which in turn will make asterisk send the fax out. I also have asterisk set up for a client where all it does is send and recieve faxes. They have 14 fax machines on SPA2000 to receive faxes and then there are 40 stations connected to ATA's to send faxes out. Of course they are using multiple data T1's connected to the internet which are very stable. - Original Message - From: "Michael K. Rodriguez" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, March 25, 2005 11:29 PM Subject: Re: [Asterisk-Users] faxes I have tested a fax call on asterisk with success. I used an IAXy on a broadband Time Warner connection. Faxes are much more sensitive than voice calls. If you have a good internet connection, faxes should complete fine. The only downfall it is recommended that you call to verify fax transmission after every fax. -Michael On 3/25/05 10:59 PM, "AS" <[EMAIL PROTECTED]> wrote: Is it possible and if so for a workstation user to send his fax via asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can I get a sip doorbell?
My home office is away from my house - so if anyone rings door I cannot hear it. How would I rig up a doorbell which would ring an extension on my Asterisk box? Angus Comber [EMAIL PROTECTED] ___ 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] What is web login password for Asteirsk@Home
Hello I have setup [EMAIL PROTECTED] and can login to the system via the asterisk box. But if I try same username and password to login using the Asterisk Management Portal I try the same username and password and cannot login. says authorization failure. I have tried from a Windows 2000 and a Windows XP machine running Internet Explorer v6. What am I doing wrong?Angus ComberItel Office Software Ltd5 Enmore GardensLondon, SW14 8RFTel: 020 8878 7367Fax: 020 8876 7257Em: [EMAIL PROTECTED]web: www.iteloffice.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
