Re: [asterisk-users] Looking for IAX trunk/DID to replace Junction Networks
On Tue, Apr 24, 2012 at 6:17 PM, John Novack jnov...@stromberg-carlson.org wrote: Voip.ms is high quality, handles number ports and supports both IAX2 and SIP 2 different pricing plans, and their costs range from 4.95 to 7.95 per month depending on the rate center for one plan, and less with no free incoming minutes for their value plan This looks like a perfect replacement that should save me a few dollars off my current price. Thanks to everyone for the suggestion, Russell. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Prevent cell phone voice mail capturing call
Hi, I've a DID number that gets passed to three internal phones and a cell phone via my outbound IAX trunk. If the cell phone is off or out of coverage, its voice mail captures the call. What's the best way to avoid this? Is there a recommended way to force the cell phone user to press 1 before the call is passed there ala google voice? Or is there another way to detect the presence of the answering machine rather than a human? Thanks, Russell. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent cell phone voice mail capturing call
On Thu, Nov 5, 2009 at 3:51 PM, Danny Nicholas da...@debsinc.com wrote: You can dial the cell like this Dial(DAHDI/1c/w5551212) instead of Dial(DAHDI/1/w5551212) Danny - thanks, however I think that's a feature of DAHDI. My outbound trunk is IAX. I don't think that's a standard feature of the dial command. Has anyone else re-implemented it for other channels? Russell. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pass CallerID when call forwards to PSTN?
On Nov 20, 2007 5:06 AM, Dovid B [EMAIL PROTECTED] wrote:
From what I have seen in the past asterisk should pass along the CID
automatically. As some one else already mentioned. It can be your ITSP. You
can always set the CID with Set(CALLERID(num)=1234567890).
Asterisk does pass the caller ID for the internal calls, but for the
external ones, my default outbound CallerID gets used.
I can set a different CID like you suggest above, but I don't know how
to get the inbound CID so I can set it correctly. Does anyone know if
there's a variable exposed to my extensions.conf so I can do something
like Set(CALLERID(num)=${VAR}) and set outbound callerID to that of
the calling party?
Thanks,
Russell
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Re: [asterisk-users] Pass CallerID when call forwards to PSTN?
On Nov 17, 2007 8:13 PM, Robert Lister [EMAIL PROTECTED] wrote: I think your carrier has to permit you to set callerID to something that is not one of your numbers in the range you have been allocated. I know I can set a different caller ID - I'm just not sure how in Asterisk I would set the callerID to be that of the number that has called me. As I mention, the call gets forwarded like this: exten = s,1,Dial(SIP/2201SIP/2202IAX2/[EMAIL PROTECTED]),90) Maybe I should be doing that another way? if not, I need a way to set the callerID before that line where the value is equal to that of the calling party. Anyone else got a suggestion? Thanks, Russell ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pass CallerID when call forwards to PSTN?
Hi, Incoming calls to one of my lines are set to ring two internal lines and simultaneously start ringing my cell phone. Something like this: exten = s,1,Dial(SIP/2201SIP/2202IAX2/[EMAIL PROTECTED]),90) The internal lines 2201 and 2202 will both see the callerID for the incoming call, but my cell phone will show the callerID for asterisk, not the calling party. What's the best solution to take the callerID from the inbound call and transfer it to the outbound one? I'm still using v1.2 here. Thanks, Russell. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gizmo project answers every call - can I use it in hunt group?
Hi, I've set up a Gizmo Project account for access on my Nokia E61 because they work through NAT. Trouble is If I include my gizmo account in an asterisk hunt group and I'm not connected (phone is off / outside wireless coverage) the gizmo project always answers. Either the call goes to voice mail or if I turn voicemail off the call gets answered by a recording saying I'm not connected. That then ends any hunt. Is there anything I can do at the asterisk end to work around this? Thanks, Russell ___ --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] Dialplan - busy and unavailable without priority jumping
Hi folks,
Moving on to a new install, I'm jumping straight to v1.4
Without using Priority jumping I'm wondering what the 'standard' way
to indicate to the calling party that the number the dialed is busy or
unavailable. So,if I have an entry in extensions.conf like this:
[outbound]
exten = _01.,1,SetCallerID(01235554321)
exten = _01.,n,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN},60)
What should I be adding to this so when a number is in use the caller
gets a busy tone, and if the call fails (i.e. the number is
unavailable) the caller gets the info tone.
All the calls will be from IAX users and be carried over an IAX trunk
to the PSTN.
Thanks,
Russell.
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Re: [asterisk-users] Dialplan - busy and unavailable without priority jumping
On 1/18/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Looks at macro-stdexten in extensions.conf.sample. Also see show application dial Ah, that's exactly what I was looking for - thanks. Russell ___ --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] Ring tone too loud on IAX channel
Hi, We are using MozIAX as a softphone with a USB headset and are making outbound calls using IAX with ulaw encoding to our voip provider. We're running asterisk 1.4 Users are complaining that the ring tone generated by asterisk is much louder than the voice call once connected. They are having to turn the volume down to avoid being deafened by the ring tone, but then have an unacceptably low volume for the voice call. Can anyone suggest what might be the problem here, or steps I could take to address it? Thanks, Russell. ___ --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] Voicemail beep doesn't end
I've hit a problem with Voicemail. My call gets answered but the 'beep' before I should start recording a message doesn't end - it gets a little quieter. I can leave a message over the top of it, but the recorded message is very quiet. Any idea what might be the cause of this problem? My config is pretty basic at the moment: [general] format=wav attach=yes [default] 101 = ,Russell Horn,[EMAIL PROTECTED] Everything else seems to work fine with inbound and outbound calling using SIP and IAX2 trunks. Thanks for any suggestions, Russell ___ --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
Re: [Asterisk-Users] No incoming sip calls
Following up to my earlier post. I'm seeing no inbound SIP traffic locally despite, apparently, being sucessfuly reigstered with my sip provider. sip show peers give me Name/username HostDyn Nat ACL Port Status 2201/2201 192.168.1.100D 5060 OK (15 ms) Gradwell/796 193.111.200.56 N 5060 OK (138 ms) 2 sip peers [2 online , 0 offline] When I look at the gradwell control panel I see that it has me registered with an IP of 192.168.1.102 - that's the internal IP of my asterisk box. Wouldn't you expect to see the external IP? sip.conf contains: bindaddr=0.0.0.0; address to bind to (0.0.0.0 binds to all) externalip=yyy.yyy.yyy.yy nat=yes ; NAT settings allow=all canreinvite=no Any pointers would be really appreciated. Thanks, Russell ___ --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] No incoming sip calls
Hi folks - I've recently returned to asterisk after an eighteen month break.
I've two sip providers - gradwell in the UK (inbound and outbound)
and talklite in the US (outbound only).
I've managed to get outbound dialing working but am not receiving any
calls from gradwell.
I've included my sip.conf and extensions.conf as well as the output
from tethereal. When a call is placed to rgadwell I'm seeing no sip
traffic whatsoever on asterisk. My aim is to have inbound calls ring
SIP extension 2201
I'm guessing this is something pretty straightforward, but any help
would be much appreciated.
Thanks,
Russell.
sip.conf
[general]
context=incoming; Default context for incoming calls
register = 7960xxx:[EMAIL PROTECTED]/2001
register = 9479xxx:[EMAIL PROTECTED]
port=5060 ; UDP Port to bind to (SIP standard
port is 5060)
bindaddr=0.0.0.0; address to bind to (0.0.0.0 binds to all)
nat=yes ; NAT settings
allow=all
[Gradwell]
type=peer
username=796
fromuser=796
secret=
host=sip.gradwell.net
context=flat
fromdomain=sip.gradwell.net
nat=yes
allow=all
canreinvite=no
dtmfmode=inband
qualify=yes
[talklite]
type=peer
username=9479
qualify=yes
secret=
host=sip.talklite.net
canreinvite=yes
disallow=all
allow=ulaw
[2201]
type=friend
context=flat
username=albanach
secret=
defaultip=192.168.1.100
qualify=yes
type=friend
callerid=Russell Horn
host=dynamic
nat=no ; X-Lite is behind a NAT router
canreinvite=yes ; Typically set to NO if behind NAT
allow=all
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
extensions.conf
[general]
static=yes
writeprotect=no
[globals]
TRUNK=Gradwell
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
PHONES1=SIP/2201
[flat]
include = home
include = outgoing
[home]
exten = 2201,1,Dial(${PHONES1},20,Ttm)
exten = 2201,2,Macro(vmessage,${PHONES1VM})
exten = 2201,3,Hangup
[outgoing]
ignorepat = 9
ignorepat = 8
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
linux:/etc/asterisk # tethereal -R sip
Capturing on eth0
0.00 207.44.248.78 - 192.168.1.102 SIP Request: OPTIONS
sip:[EMAIL PROTECTED]
0.000831 192.168.1.102 - 207.44.248.78 SIP Status: 404 Not Found
1.350584 192.168.1.102 - 192.168.1.100 SIP Request: OPTIONS
sip:[EMAIL PROTECTED]:5060
1.350730 192.168.1.102 - 207.44.248.78 SIP Request: OPTIONS
sip:sip.talklite.net
1.350887 192.168.1.102 - 193.111.200.56 SIP Request: OPTIONS
sip:sip.gradwell.net
1.369388 192.168.1.100 - 192.168.1.102 SIP Status: 200 OK
1.455492 207.44.248.78 - 192.168.1.102 SIP Status: 404 Not Found
1.502618 193.111.200.56 - 192.168.1.102 SIP Status: 404 Invalid
account for voicemail
1.552845 192.168.1.102 - 207.44.248.78 SIP Request: REGISTER
sip:sip.talklite.net
1.654933 207.44.248.78 - 192.168.1.102 SIP Status: 100 Trying(1 bindings)
1.655832 192.168.1.102 - 193.111.200.56 SIP Request: REGISTER
sip:sip.gradwell.net
1.657951 207.44.248.78 - 192.168.1.102 SIP Status: 401 Unauthorized
(1 bindings)
1.658229 192.168.1.102 - 207.44.248.78 SIP Request: REGISTER
sip:sip.talklite.net
1.770875 207.44.248.78 - 192.168.1.102 SIP Status: 100 Trying(1 bindings)
1.773894 207.44.248.78 - 192.168.1.102 SIP Status: 200 OK(1 bindings)
1.792718 193.111.200.56 - 192.168.1.102 SIP Status: 401
Unauthorized(0 bindings)
1.793529 192.168.1.102 - 193.111.200.56 SIP Request: REGISTER
sip:sip.gradwell.net
1.937253 193.111.200.56 - 192.168.1.102 SIP Status: 200 OK(1 bindings)
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Re: [Asterisk-Users] No incoming sip calls
On 6/13/06, Jonathan Attwood [EMAIL PROTECTED] wrote: Could your register line require attention ? (2001?) 7960xxx:[EMAIL PROTECTED]/2001 - I thought your target was 2201? That's a good spot and I've fixed it now, but I'm sure it's not the problem. I'm not seeing any sip traffic coming in at all, I'd have expected if I jsut had the wrong extensions to have seen both traffic and errors at the console. Thanks though - I'll keep looking. Russell ___ --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] Braodvoice - UK Non Geographic Numbers
Since May 05 I have been unable to call any non-geographic number in the UK via Broadvoice. Thse are numbers such as the 0800 range (free to call) 087xx (local / national rate calls). Broadvoice support have been unhelpful, and can't say if there's any intention to fix this. A case has been upen since May 24 without any updates. Is anyone else having this problem? Has anyone else spoken to broadvoice about it? Did you get any further? Is there any indication it might be resolved? The last customer rep I spoke to recommended I close my account if I need to dial these numbers - I'd prefer to keep my phone number, but if all else fails... Russell. ___ 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] Re: Braodvoice - UK Non Geographic Numbers
Broadvoice could connect to non geographic numbers without difficulty until the fourth week of May 2005. I can call non-geographic numbers from my land line in the US, my mobile phone and from any calling card I have tried. This isn't an issue with BT but with broadvoice and those they contract to supply connections to the UK PSTN. On 7/7/05, Michael Welter [EMAIL PROTECTED] wrote: Russell Horn wrote: Since May 05 I have been unable to call any non-geographic number in the UK via Broadvoice. Thse are numbers such as the 0800 range (free to call) 087xx (local / national rate calls). Broadvoice support have been unhelpful, and can't say if there's any intention to fix this. A case has been upen since May 24 without any updates. Is anyone else having this problem? Has anyone else spoken to broadvoice about it? Did you get any further? Is there any indication it might be resolved? The last customer rep I spoke to recommended I close my account if I need to dial these numbers - I'd prefer to keep my phone number, but if all else fails... Russell. ___ 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 I lost a client because of this. BT will not allow premium numbers to be called from outside the UK. I even tried it from an ITSP in the Netherlands, and the call didn't go through :-( The ATT monopoly is gone. Hopefully, BT's time will come--the sooner the better. ___ 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] ANNOUNCEMENT : NEW CallingCard Application forAsterisk
It is better to stay with Postgres. If you don't want to loose your business stay away from MySQL. Oh come on, there are many reasons to use Postgres, but this is just FUD. Just as an example off the top of my head, take a look at http://www.livejournal.com/stats.bml (2.5 million active accounts, 367,000 updates in the last 24 hours and all on a mysql backend). There's a host of other big sites all using MySQL - Yahoo! Finance, Slashdot (handling 360 queries per second) and others. If you're losing data on MySQL with 10 users you have a configuration or coding problem. Again, Postgres offers many features that MySQL does not and vice versa, but to suggest that MySQL shouldn't be used because you'll loose data is a bogus argument. Russell. ___ 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] Problem ringing simultaneous channels
I have a problem with ringing simultaneous channels where one is IAX and one is Zap I have two Zap channels and a single extensions on IAX2 I'm trying to take incoming calls on Zap/1 and if not answered in 15 seconds by IAX2/100 to keep ringing IAX2 and also try another number on Zap/2 Unfortunately it seems that when asterisk tries to ring the other number on Zap/2 it thinks the call has been answered and can therefore stop ringing IAX2 Here's what I have in extensions: [officeopen] exten = s,1,Dial(IAX2/100,15) exten = s,2,Dial(IAX2/100Zap/2/07879xx,15) exten = s,3,Playback(nooneavailable) exten = s,4,Voicemail2(u2000) exten = s,5,Hangup And here's the output from asterisk: -- Starting simple switch on 'Zap/1-1' -- Executing Dial(Zap/1-1, IAX2/100|15) in new stack -- Called 100 -- Call accepted by 67.76.xxx.xxx (format GSM) -- Format for call is GSM -- IAX2/100/1 is ringing -- Nobody picked up in 15000 ms -- Hungup 'IAX2/100/1' -- Executing Dial(Zap/1-1, IAX2/100Zap/2/07879xx|15) in new stack -- Called 100 -- Called 2/07879xx -- Call accepted by 67.76.162.215 (format GSM) -- Format for call is GSM -- IAX2/100/3 is ringing -- Zap/2-1 answered Zap/1-1 -- Hungup 'IAX2/100/3' -- Attempting native bridge of Zap/1-1 and Zap/2-1 -- Hungup 'Zap/2-1' == Spawn extension (from-analog-a, s, 2) exited non-zero on 'Zap/1-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (from-analog-a, h, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Here's hoping someone can help! Many thanks, Russell. ___ 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] Problem ringing simultaneous channels
Alexander, I'm afraid it's POTS (X101P) and from what I have seen since I posted this is my problem. I wouldn't mind it hanging up the IAX2 channel and then calling it again, but it seems that everytime the new call via Zap/2 means no other calls are possible. There is ISDN in the office, but I don't have any access until April :/ If what I'm trying is impossible it will just have to wait Russell. ___ 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] Broadvoice Problems
Has anyone else encountered Broadvoice problems today? I was unable to log in at all until after lunchtime. Now I can connect but any calls ring once then get 480 Temporarily Unavailable back from Broadvoice. It's now also impossible to call their support desk, any calls receive a recorded message saying the mailbox is full and then get hung up. Just wondering if I'm alone in this? Russell. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice problem
Since Thursday evening my asterisk box has been failing to register with broadvoice. I haven't changed any of my config files in the last week. Can anyone suggest anything? Asterisk is reporting: *CLI Aug 28 16:15:17 NOTICE[6150]: chan_sip.c:3914 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again -- Got SIP response 404 Not found back from 147.135.8.129 Urgent handler My broadvoice config in sip.conf looks like: [general] context=incoming; Default context for incoming calls externalip=82.41.201.XXX register = 703XXX:[EMAIL PROTECTED] port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; address to bind to (0.0.0.0 binds to all) srvlookup=no; Enable DNS SRV lookups on outbound calls [Broadvoice] type=peer username=703XXX fromuser=703XXX secret=PASSWORD host=147.135.8.129 context=flat fromdomain=147.135.8.129 nat=no disallow=all allow=ulaw canreinvite=no dtmfmode=inband qualify=yes tethereal -V port 5060 reports: Frame 11 (416 on wire, 416 captured) Arrival Time: Aug 28, 2004 16:17:05.72973 Time delta from previous packet: 4.093142000 seconds Time relative to first packet: 20.001957000 seconds Frame Number: 11 Packet Length: 416 bytes Capture Length: 416 bytes Ethernet II Destination: 00:0d:66:23:84:54 (00:0d:66:23:84:54) Source: 00:40:95:35:d0:b8 (R.P.T._35:d0:b8) Type: IP (0x0800) Internet Protocol, Src Addr: 82-41-201-.cable.ubr11.edin.blueyonder.co.uk (82.41.201.160), Dst Addr: 147.135.8.129 (147.135.8.129) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x10 (DSCP 0x04: Unknown DSCP; ECN: 0x00) 0001 00.. = Differentiated Services Codepoint: Unknown (0x04) ..0. = ECN-Capable Transport (ECT): 0 ...0 = ECN-CE: 0 Total Length: 402 Identification: 0x000d Flags: 0x04 .1.. = Don't fragment: Set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 64 Protocol: UDP (0x11) Header checksum: 0x816c (correct) Source: 82-41-201.cable.ubr11.edin.blueyonder.co.uk Destination: 147.135.8.129 (147.135.8.129) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Source port: sip (5060) Destination port: sip (5060) Length: 382 Checksum: 0x6327 (correct) Session Initiation Protocol Request line: REGISTER sip:147.135.8.129 SIP/2.0 Message Header Via: SIP/2.0/UDP 82.41.201.160:5060;branch=z9hG4bK30718407 From: sip:[EMAIL PROTECTED];tag=as38aec91c To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 106 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 Frame 12 (348 on wire, 348 captured) Arrival Time: Aug 28, 2004 16:17:05.995393000 Time delta from previous packet: 0.265663000 seconds Time relative to first packet: 20.26762 seconds Frame Number: 12 Packet Length: 348 bytes Capture Length: 348 bytes Ethernet II Destination: 00:40:95:35:d0:b8 (R.P.T._35:d0:b8) Source: 00:0d:66:23:84:70 (00:0d:66:23:84:70) Type: IP (0x0800) Internet Protocol, Src Addr: 147.135.8.128 (147.135.8.128), Dst Addr: 82-41-201-XXX.cable.ubr11.edin.blueyonder.co.uk (82.41.201.XXX) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00) 00.. = Differentiated Services Codepoint: Default (0x00) ..0. = ECN-Capable Transport (ECT): 0 ...0 = ECN-CE: 0 Total Length: 334 Identification: 0xf020 Flags: 0x00 .0.. = Don't fragment: Not set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 49 Protocol: UDP (0x11) Header checksum: 0xe0ad (correct) Source: 147.135.8.128 (147.135.8.128) Destination: 82-41-201-XXX.cable.ubr11.edin.blueyonder.co.uk (82.41.201.XXX) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Source port: sip (5060) Destination port: sip (5060) Length: 314 Checksum: 0x15e1 (correct) Session Initiation Protocol Status line: SIP/2.0 404 Not found Message Header Via: SIP/2.0/UDP 82.41.201.XXX:5060;branch=z9hG4bK30718407 From: sip:[EMAIL PROTECTED];tag=as38aec91c To: sip:[EMAIL PROTECTED];tag=SD30va299-239804385-1093709825857 Call-ID: [EMAIL PROTECTED] CSeq: 106 REGISTER Content-Length: 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Remotely change call forward
Is it possible using asterisk to allow someone to dial in and remotely change where their call is forwarded to? For example, I'm working from home so I want my calls to go to 555 1234, now I need to go out for a bit so I'd like to phone the office and using DTMF tell the asterisk PBX to now forward my calls to my cell phone 555 3456 Has anyone implimented anything like this? R. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with ougoing Zap calls
I'm able to receive but not make calls with zaptel using an X101P
connecting to Asterisk with an Xlite client. My client has context = flat
in sip.conf and extensions number 8919
In extensions.conf I've got:
[home]
; Line 1
;
exten = 8919,1,Dial(${PHONES1},20,Ttm)
exten = 8919,2,Macro(vmessage,${PHONES1VM})
exten = 8919,3,Hangup
[outgoing]
exten = _9.,1,Dial(Zap/1/$EXTEN:1)
[flat]
include = home
include = outgoing
zapata.conf contains the following - I have 2 x101p cards installed
[channels]
language=en
group=1
context=from-analog
signalling=fxs_ks
usecallerid=no
echocancel=yes
echocancelwhenbridged=yes
channel = 1-2
When I dial the asterisk box from an ordinary phone it picks up fine and
shows:
-- Starting simple switch on 'Zap/1-1'
-- Executing Dial(Zap/1-1, SIP/8919|30) in new stack
-- Called 8919
-- SIP/8919-2e9f is ringing
But if I try and dial out I get:
-- Executing Dial(SIP/8919-917c, Zap/1/$EXTEN:1) in new stack
-- Called 1/$EXTEN:1
-- Zap/1-1 answered SIP/8919-917c
-- Hungup 'Zap/1-1'
== Spawn extension (flat, 95558925, 1) exited non-zero on 'SIP/8919-917c'
So my call is immediatly answered, but doesn't go anywhere, then asterisk
hangs up.
Has anyone else experienced this, or know what's wrong? Help would be
very much appreciated!
Russell.
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