Re: [Asterisk-Users] Development news :: T38 passthrough support
The sipura 2100 does work good with a AS5300 Zoa wrote: Does anybody know what devices really support t.38 ? I've seen a few claiming they do on the box, but most do not seem to support it at all. Zoa. Kristian Kielhofner wrote: Olle E Johansson wrote: Friends in the Asterisk.org community, There is a lot of cool stuff going on in Asterisk development, things that will change Asterisk and make it work better in your organisation, make it easier to sell in your area or give you more consulting oppurtunities - in short, functionality that will make a lot of sense for you users. However, developers can't really get anywhere without a dialog with the users. You know what you need, you know what is missing and how you would like to make Asterisk a better choice. I am planning to send out a description of new features now and then, to inform you about what is going on, but also to get some feedback. The bug tracker is not only a tool for developers, but also for testers and users to react to changes and contribute. *** ITU T.38 -- Fax over VoIP Olle, Let's say that I wanted to setup a complete environment to test this. I presume that I would need the following: Fax machine T.38 compliant ATA (Sipura claims this) Asterisk server T.38 compliant something - does this need to be a Cisco 5300 (or similar)? Can it be just another plain ATA and fax machine? Please suggest some possible hardware! Thanks! ___ --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] spandsp and page orientation
Shawn, you ever get a fix for this problem? samples are at http://tumtum.no-ip.com/faxes/1128432831.3.tif http://tumtum.no-ip.com/faxes/853107320051004-150908.tif Both of these were faxed from a Brother intellifax 750 through a ring-it single-line simulator into my asterisk box (through an X100P clone) both were normal 8.5X11 pages in portrait style (the map image should be 8.5 wide and 11 long) I can't take the old fax machine offline until I get this resolved. If anyone has any ideas I am open to suggestion. Shawn -Original Message- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Shawn Porter Sent: Tuesday, October 04, 2005 10:32 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] spandsp and page orientation I have just installed spandsp-0.0.2 onto my Asterisk 1.0.9 I am using an old Intel 536EP (actually found drivers that work) BUT...all my faxes are coming in landscape mode Has anyone come across this? any fixes? Shawn ___ --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] Asterisk as a Gateway
Nitesh Divecha wrote:
Are there any examples of dial plans? Like how to make the default
context?
I just need a kick start on the config part, as I am really struggling
on routing the calls.
Here is a very very simple example using a PRI. You will need more error
routing in a real dial plan:
extensions.conf:
[general]
static=yes
writeprotect=no
country=us
[local]
include = default
[globals]
TRUNK=Zap/g1
LDTRUNK=Zap/g2
[trunk]
;Long distance pstn
exten = _1NXXNXX,1,Dial(${LDTRUNK}/${EXTEN})
exten = _1NXXNXX,2,Hangup
;pstn
exten = _X.,1,Dial(${TRUNK}/${EXTEN})
exten = _X.,2,Hangup
[default-out]
;This is where you sent trusted calls from sip.conf out to pstn
include = trunk
[default]
;you send incoming pstn calls here as well as untrusted voip calls.
;here you would route call to local numbers you own via enum or static.
exten = 6153247060,1,Wait(2) ; you need to wait
; long enough to get
; CNAM off line
;send incoming call to your register server.
exten = 55,2,Dial(SIP/[EMAIL PROTECTED])
sip.conf:
[general]
bindport = 5060
bindaddr = 0.0.0.0
context = default ; non trusted call from sip side go here
srvlookup = yes
dtmfmode=info
disallow=all
allow=ulaw
allow=alaw
allow=g729
[trusted]
type=friend
context=default-out ; trusted call can go out pstn
host=192.168.0.1
canreinvite=no
zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
span=3,1,0,esf,b8zs
bchan=49-71
dchan=72
span=4,1,0,esf,b8zs
bchan=73-95
dchan=96
loadzone = us
defaultzone=us
zapata.conf:
[channels]
context=default;pstn incoming call go here
switchtype=national
signalling=pri_cpe
toneduration=500
usecallerid=yes
hidecallerid=no
callwaitingcallerid=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=-1.0
txgain=-1.0
callerid=asreceived
;
group=1
channel=1-23
channel=73-95
;
group=2
channel=25-47
channel=49-71
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Re: [Asterisk-Users] Asterisk as a Gateway
See the message I post right before this one for a simple example. Ray Yang wrote: Apart from the dial plan issue, can anyone let Asterisk act like Cisco GW to accept SIP call without registered in advance? I've tried this for a long time but no answer yet. ___ --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] Asterisk as a Gateway
The line that reads:
exten = 6153247060,1,Wait(2)
should have been:
exten = 55,1,Wait(2)
Nitesh Divecha wrote:
Thanks James,
That should help to start my project Thanks a million...
I will keep on updating..
And thanks to all for the inputs
Thanks,
Neal
On Dec 29, 2005, at 6:39 AM, James Sizemore wrote:
Nitesh Divecha wrote:
Are there any examples of dial plans? Like how to make the default
context?
I just need a kick start on the config part, as I am really
struggling
on routing the calls.
Here is a very very simple example using a PRI. You will need more
error routing in a real dial plan:
extensions.conf:
[general]
static=yes
writeprotect=no
country=us
[local]
include = default
[globals]
TRUNK=Zap/g1
LDTRUNK=Zap/g2
[trunk]
;Long distance pstn
exten = _1NXXNXX,1,Dial(${LDTRUNK}/${EXTEN})
exten = _1NXXNXX,2,Hangup
;pstn
exten = _X.,1,Dial(${TRUNK}/${EXTEN})
exten = _X.,2,Hangup
[default-out]
;This is where you sent trusted calls from sip.conf out to pstn
include = trunk
[default]
;you send incoming pstn calls here as well as untrusted voip calls.
;here you would route call to local numbers you own via enum or static.
exten = 6153247060,1,Wait(2) ; you need to wait
; long enough to get
; CNAM off line
;send incoming call to your register server.
exten = 55,2,Dial(SIP/[EMAIL PROTECTED])
sip.conf:
[general]
bindport = 5060
bindaddr = 0.0.0.0
context = default ; non trusted call from sip side go here
srvlookup = yes
dtmfmode=info
disallow=all
allow=ulaw
allow=alaw
allow=g729
[trusted]
type=friend
context=default-out ; trusted call can go out pstn
host=192.168.0.1
canreinvite=no
zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
span=3,1,0,esf,b8zs
bchan=49-71
dchan=72
span=4,1,0,esf,b8zs
bchan=73-95
dchan=96
loadzone = us
defaultzone=us
zapata.conf:
[channels]
context=default;pstn incoming call go here
switchtype=national
signalling=pri_cpe
toneduration=500
usecallerid=yes
hidecallerid=no
callwaitingcallerid=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=-1.0
txgain=-1.0
callerid=asreceived
;
group=1
channel=1-23
channel=73-95
;
group=2
channel=25-47
channel=49-71
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Nitesh Divecha
VoIP/Network Engineer
Viper Networks
10373 Roselle St. Ste:170
San Diego, CA. 92121
Phone: 858-452-8737
Fax: 858-452-8638
Cell: 1-909-964-5181
vPhone: 544-416-0067
Email: [EMAIL PROTECTED]
Web: www.vipernetworks.com
Your Internet Phone Company
A publicly traded Company, OTC: VPER
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Re: [Asterisk-Users] Asterisk as a Gateway
Yes, asterisk makes a better voip to pstn gateway then Cisco. Asterisk has more advanced call routing and restrictions then Cisco gear. Nitesh Divecha wrote: Hello, Is it possible to implement Asterisk as a Gateway? For example like Cisco 5300 or 5400 with 4 T1. I was planning to buy Digium 4 port T1 card and make Asterisk as a Gateway, which will do the call routing. Any ideas? Thanks, Neal ___ --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] Asterisk does not handle call from a Cisco IAD correctly
when my Cisco IAD send a call to my Asterisk gateway the gateway treats it as if I don't have a peer statement in sip.conf, when I do. Here are the first two packets, notice the Found no matching peer or user for '192.168.7.250:50437' on the second packet. Any one seen this before, or have a clue as to the problem? Asterisk 1.0.9 sip.conf: [bna-vonx-iad] type=friend context=trusted-out host=192.168.7.250 canreinvite=no Sip read: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.250:5060;branch=z9hG4bK1A60 From: James Sizemore sip:[EMAIL PROTECTED];tag=19D8A640-5E9 To: sip:[EMAIL PROTECTED] Date: Wed, 06 Mar 2002 00:27:08 GMT Call-ID: [EMAIL PROTECTED] Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 3128236623-802099670-2154346748-2004044536 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: James Sizemore sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off Timestamp: 1015374428 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 191 v=0 o=CiscoSystemsSIP-GW-UserAgent 6047 8216 IN IP4 192.168.7.250 s=SIP Call c=IN IP4 192.168.7.250 t=0 0 m=audio 16434 RTP/AVP 0 c=IN IP4 192.168.7.250 a=rtpmap:0 PCMU/8000 a=ptime:20 20 headers, 9 lines Using latest request as basis request Sending to 192.168.7.250 : 5060 (non-NAT) Found no matching peer or user for '192.168.7.250:50437' Found RTP audio format 0 Peer audio RTP is at port 192.168.7.250:16434 Found description format PCMU Capabilities: us - 0x78e (gsm|ulaw|alaw|lpc10|g729|speex|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 615917 in default list_route: hop: sip:[EMAIL PROTECTED]:5060 Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.7.250:5060;branch=z9hG4bK1A60 From: James Sizemore sip:[EMAIL PROTECTED];tag=19D8A640-5E9 To: sip:[EMAIL PROTECTED];tag=as43478a8a Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Memphis ISDN-NET PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ___ --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] Asterisk does not handle call from a Cisco IAD correctly
I think I found what is munging up the peer lookup: This call from another Asterisk box starts: -- SIP read from 192.168.69.254:5060: The peer lookup that fail reads: -- SIP read from 192.168.7.250:52141: Asterisk seem to be thrown off by the fact that the return port is not 5060, and fails the peer lookup. This is a * bug then. I have documented it with both 1.0.9 and 1.2.1. Time to dig through the sip code. James Sizemore wrote: when my Cisco IAD send a call to my Asterisk gateway the gateway treats it as if I don't have a peer statement in sip.conf, when I do. Here are the first two packets, notice the Found no matching peer or user for '192.168.7.250:50437' on the second packet. Any one seen this before, or have a clue as to the problem? Asterisk 1.0.9 sip.conf: [bna-vonx-iad] type=friend context=trusted-out host=192.168.7.250 canreinvite=no Sip read: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.7.250:5060;branch=z9hG4bK1A60 From: James Sizemore sip:[EMAIL PROTECTED];tag=19D8A640-5E9 To: sip:[EMAIL PROTECTED] Date: Wed, 06 Mar 2002 00:27:08 GMT Call-ID: [EMAIL PROTECTED] Supported: 100rel,timer Min-SE: 1800 Cisco-Guid: 3128236623-802099670-2154346748-2004044536 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: James Sizemore sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off Timestamp: 1015374428 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 191 v=0 o=CiscoSystemsSIP-GW-UserAgent 6047 8216 IN IP4 192.168.7.250 s=SIP Call c=IN IP4 192.168.7.250 t=0 0 m=audio 16434 RTP/AVP 0 c=IN IP4 192.168.7.250 a=rtpmap:0 PCMU/8000 a=ptime:20 20 headers, 9 lines Using latest request as basis request Sending to 192.168.7.250 : 5060 (non-NAT) Found no matching peer or user for '192.168.7.250:50437' Found RTP audio format 0 Peer audio RTP is at port 192.168.7.250:16434 Found description format PCMU Capabilities: us - 0x78e (gsm|ulaw|alaw|lpc10|g729|speex|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 615917 in default list_route: hop: sip:[EMAIL PROTECTED]:5060 Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.7.250:5060;branch=z9hG4bK1A60 From: James Sizemore sip:[EMAIL PROTECTED];tag=19D8A640-5E9 To: sip:[EMAIL PROTECTED];tag=as43478a8a Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Memphis ISDN-NET PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ___ --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] How does DTMF get sent over a PRI in Asterisk
I am trying to trouble shoot some problems with DTMF over PRI. I have a digium wct1xxp card and these lines in extensions.conf: exten = 5556000,1,Record(testtone:gsm) exten = 5556000,2,Wait(2) exten = 5556000,3,Playback(testtone) I called in over the PSTN --to-- Asterisk. I did a pri debug, from asterisk 1.2.0 console and a few debug lines came up when the call connects (but not all the q931 message that should have been there). But nothing came up when hitting the dtmf keys. Also when I listen to the play back the recorded tones are little more then chirps not long enough for a human ear to distinguish tone. My two question are how does DTMF get send over a PRI (inband? q931 message?) and how would I go about seeing the duration that was sent to me so that I know weather the problem is Asterisk or my telcos! ___ --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] suggestions for hard phones?
Does the SPA-941 support stun? Kerry Garrison wrote: My two favorite phones (in order) are: Linksys SPA-941 http://voipspeak.net/index.php?option=com_contenttask=viewid=41 Grandstream GXP-2000 http://voipspeak.net/index.php?option=com_contenttask=viewid=25 The problem is the change of credentials, that is an interesting issue. With either phone, you can have multiple accounts assigned to it and the user can set the DO-Not-Disturb for their line when they come and go. That is probably the easiest way to accomplish it. The second would be a single extension for the actual phone and then a call queue for each person. When each person comes into work, they log into their own call queue. The first approach is easier to implement. Kerry Garrison http://voipspeak.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Fraser Sent: Thursday, November 17, 2005 3:44 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] suggestions for hard phones? Hi all, I am looking for SIP hard phones to use in a call center. The feature that I need the most is quick change of logon credentials as we run 3 shifts. each agent will have their own extension number and password. any suggestions would be greatly appreciated. thank you John Fraser ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.3/173 - Release Date: 11/16/2005 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.3/173 - Release Date: 11/16/2005 ___ --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] Info on beta1 seem to be broke
I have a beta1 gateway with a 4 port card in PRI mode, the gateway sends DTMF via sip info packets to another beta1 box. The peer is set to receive info. What I get is a click sound and a very very short tone. Sound like to me that I get the first part of the tone before it is captured and put in the info packet, but the gateway never seems to send the tone, the packet that gets sent looks like this: -- -- SIP read from 192.168.117.4:5060: INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.117.4:5060;branch=z9hG4bK7bd677b8 From: WIRELESS CALLER sip:[EMAIL PROTECTED];tag=as37dd610c To: sip:[EMAIL PROTECTED];tag=as3af9dc41 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 120 INFO User-Agent: ISDN-NET Voip Gateway Content-Type: application/dtmf-relay Content-Length: 24 Signal=5 Duration=500 --- I know it is not the receiving box messing things up as I get the same short short DTMF sound on a cisco IAD. Something is wrong with this packet but I just can't see it!!! Is there any rtp that gets sent, anyone know what the Content-length does? ___ --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] Info on beta1 seem to be broke
After reading through the rfc http://www.ietf.org/rfc/rfc2976 and cisco site http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftinfo.htm And googling a bit I can not find anything that look out of place, I notices a few formating difference on examples I have seen and most examples have a content-length: 26 but I don't think that should matter, and the rfc does not look like there is any rtp needed for info application/dtmf-relay packets. Could if be as simple and the space after the = INFO sip:201 at 192.168.1.38 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.36:5060 From: sip:101 at 192.168.1.36;tag=43 To: sip:201 at 192.168.1.38;tag=9753 Call-ID: 100450864100 at 192.168.1.36 CSeq: 3 INFO Content-Length: 26 Content-Type: application/dtmf-relay Signal= 2 Duration= 110 James Sizemore wrote: I have a beta1 gateway with a 4 port card in PRI mode, the gateway sends DTMF via sip info packets to another beta1 box. The peer is set to receive info. What I get is a click sound and a very very short tone. Sound like to me that I get the first part of the tone before it is captured and put in the info packet, but the gateway never seems to send the tone, the packet that gets sent looks like this: -- -- SIP read from 192.168.117.4:5060: INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.117.4:5060;branch=z9hG4bK7bd677b8 From: WIRELESS CALLER sip:[EMAIL PROTECTED];tag=as37dd610c To: sip:[EMAIL PROTECTED];tag=as3af9dc41 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 120 INFO User-Agent: ISDN-NET Voip Gateway Content-Type: application/dtmf-relay Content-Length: 24 Signal=5 Duration=500 --- I know it is not the receiving box messing things up as I get the same short short DTMF sound on a cisco IAD. Something is wrong with this packet but I just can't see it!!! Is there any rtp that gets sent, anyone know what the Content-length does? ___ --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] chan_zap ignoring stuff in beta1?
I just upgraded to beta1 and everything does seem to be working, however when reloading asterisk I see these error messages: -- Reloading module 'chan_zap.so' (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Oct 29 20:33:13 WARNING[10141]: chan_zap.c:10593 setup_zap: Ignoring switchtype Oct 29 20:33:13 WARNING[10141]: chan_zap.c:10593 setup_zap: Ignoring signalling Oct 29 20:33:13 WARNING[10141]: chan_zap.c:10593 setup_zap: Ignoring toneduration Now the pri's do load and are signaling via national 2 but I would like to know why they are being ignored and how do I get it to not Ignore tone duration? ___ --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] INFO Duration=250
Where can I change the Duration length of an INFO packet? Content-Type: application/dtmf-relay Content-Length: 24 Signal=5 Duration=250 ___ --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] Snom phones?
They are good phones they allow you to use speed-dial and hints to one button park and intercom. The only bad thing I can say about them is that there is no ground loop detection when using cinch headsets. I only ever had one user complaint with the Snom phones, some users don't like how soft the side tone is, and it can not be adjusted. All and all I like them better then Cisco 7960/40(no support for stun and zero echo suppression) and Grandstream GPX-2000 (good features but can be flaky) Stephen Bosch wrote: Hi, everyone: I'm in the processing of deciding what IP phones we should use with our Asterisk server, and I wanted to get feedback from the user community on the quality, reliability and ease of operation of Snom phones. What do you have to say about these phones? Are there other phones you'd suggest along with or instead of Snom? Thanks, -Stephen- ___ --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] Distinctive ringing on Cisco 79xx
exten = s,1,SetVar(ALERT_INFO=Bellcore-dr4) Bruce Komito wrote: Greetings, I am trying to implement distinctive ringing on a Cisco 7960. I have tried setting alert_info to chirp1 or chirp2 before dialing the phone, but it has no affect. If you have successfully implemented distinctive ringing on a 7960, I would really appreciate seeing the snipit of code that works. Thanks in advance Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ --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 x PROLIANT ML 150 G2 SATA
Fedora core 3 supports SATA on that model. listas iPfone wrote: Hi All, I´m tryingo to install asterisk in an PROLIANT ML 150 G2 SATA and can´t make it work because linux cant recognize the Hd (HP 160 mb). No drivers for Centos ...Red Hat... i´t´s drivig me crazy.. Someone have a tip? if i make change it to SCSI i think it will work but not sure about. Thanks Miklos ___ 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] PRI Cause Code Help
; PRI Out of band indications. ; Enable this to report Busy and Congestion on a PRI using out-of-band ; notification. Inband indication, as used by Asterisk doesn't seem to ;work ; with all telcos. ; outofband: Signal Busy/Congestion out of band with ;RELEASE/DISCONNECT ; inband: Signal Busy/Congestion using in-band tones priindication = outofband Trevor Peirce wrote: Hello, I just got off the phone with my PRI provider, and was told that I am not sending an expected message when I reject a call with a Cause Code for Unassigned(1) and Congestion (42). Busy works fine though... Anyhow, they are seeing the RELEASE COMPLETE message with cause code 1, however the tech told me they expect a PROGRESS indicator with a value between 1 and 10. Any ideas on how to resolve this? Thanks, Trevor Peirce ___ 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] PRI Card TE110p Question
Install a smp kernel and you will use IO-APIC instead of XT-PIC you typically will not share interrupts in APIC mode because it has twice the numbers of interrupts to use. Ronald Hartmann wrote: Good Day list, I am having some issues with my card in that it wants to share IRQs with everything else in my box. I am running WhiteBox Linux 3.0 Is there a way to tell linux to assign a specific IRQ to a card. (unfortunately my MB does not have feature of assigning IRQ to slot) Further, I know this is silly to even ask but I have to put my rear-end out there and say should I have bios set for PNP yes or No Thanks for your help ~ron ___ 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 Not hanging up DS0 when number called is busy.
I have a PRI that if you dial a number that is busy, the channel does
not hang up, it then sends h|1to the phone company which will then
plays back to the end sip user You don't need to dial a one or zero
I am running stable CVS-v1-0-01/20/05-02:45:17. I have placed the
important bit from the extension and sip configs below. Simplest
possible example that will show the problem. Anyone run into this
problem before?
-- Executing
Dial(SIP/192.168.69.254-08d76480,Zap/g1/5554441133) in new stack
-- Called g1/5554441133
-- Channel 0/1, span 1 got hangup
-- Zap/1-1 is busy
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time
-- Timeout on SIP/192.168.69.254-08d76480
== CDR updated on SIP/192.168.69.254-08d76480
-- Executing Goto(SIP/192.168.69.254-08d76480, h|1) in new stack
-- Goto (default-out,h,1)
-- Executing Hangup(SIP/192.168.69.254-08d76480, ) in new stack
== Spawn extension (default-out, h, 1) exited non-zero on
'SIP/192.168.69.254-08d76480'
-- Executing Hangup(SIP/192.168.69.254-08d76480, ) in new stack
== Spawn extension (default-out, h, 1) exited non-zero on
'SIP/192.168.69.254-08d76480'
extensions.conf:
[trunk]
exten = _X.,1,Dial(${TRUNK}/${EXTEN})
exten = h,1,Hangup
[default-out]
include = trunk
sip.conf:
[office]
type=friend
host=192.168.69.254
context=default-out
canreinvite=no
dtmfmode=inband
accountcode=office
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Re: [Asterisk-Users] Asterisk Not hanging up DS0 when number called is busy.
Already did.
Thanks for taking a stab though.
Brancaleoni Matteo wrote:
hi,
Il giorno mar, 01-02-2005 alle 13:30 -0600, James Sizemore ha scritto:
extensions.conf:
[trunk]
exten = _X.,1,Dial(${TRUNK}/${EXTEN})
exten = h,1,Hangup
try
extensions.conf:
[trunk]
exten = _X.,1,Dial(${TRUNK}/${EXTEN})
exten = _X.,2,Hangup
Matteo.
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[Asterisk-Users] PRI not hanging up the channel after Executing Hangup when dialing busy number.
I have a PRI that if you dial a number that is busy, the channel does
not hang up, it then sends h|1to the phone company which will then
plays back to the end sip user You don't need to dial a one or zero
I am running stable CVS-v1-0-01/20/05-02:45:17. I have pasted the
important bit from the exten and sip configs below simplest possible
example that will show the problem.
Anyone run into this problem before?
-- Executing
Dial(SIP/192.168.69.254-08d76480,Zap/g1/5554441133) in new stack
-- Called g1/5554441133
-- Channel 0/1, span 1 got hangup
-- Zap/1-1 is busy
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time
-- Timeout on SIP/192.168.69.254-08d76480
== CDR updated on SIP/192.168.69.254-08d76480
-- Executing Goto(SIP/192.168.69.254-08d76480, h|1) in new stack
-- Goto (default-out,h,1)
-- Executing Hangup(SIP/192.168.69.254-08d76480, ) in new stack
== Spawn extension (default-out, h, 1) exited non-zero on
'SIP/192.168.69.254-08d76480'
-- Executing Hangup(SIP/192.168.69.254-08d76480, ) in new stack
== Spawn extension (default-out, h, 1) exited non-zero on
'SIP/192.168.69.254-08d76480'
extensions.conf:
[trunk]
exten = _X.,1,Dial(${TRUNK}/${EXTEN})
exten = h,1,Hangup
[default-out]
include = trunk
sip.conf:
[office]
type=friend
host=192.168.69.254
context=default-out
canreinvite=no
dtmfmode=inband
accountcode=office
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Re: [Asterisk-Users] T100P frame slips
Try commenting out ;echocancel=yes ;echotraining=yes I bet your faxs start working in both directions. But of course you will now have occasional echo problems. Andrew Kohlsmith wrote: On December 23, 2004 08:29 pm, Steve Underwood wrote: This point is interesting. On most systems, if you cannot here regular ticks its pretty certain there are no slips. With the Digium cards, for some reason, many people have slips (usually due to configuration issues rather than faulty cards) yet were unaware of this until they have trouble with my spandsp software. That is weird. I am receiving faxes with no issues at all now using spandsp and app_rxfax, however passing the same audio stream on to a real fax machine (Canon IR3300 and/or some manner of ancient laser fax which does support 9600 baud) causes bad receives. This is on a TDM430P, and was tested in a Xeon 2.8 Supermicro motherboard and also on a rinky-dink old P3-800 (app_rxfax works just great on the latter and is in fact what we're using). Want to know the kicker? sending faxes out from either of those fax machines mentioned above through the exact same TDM430P... no issues at all. Just receiving. A T100P + Adit600 channel bank had no problems with those same faxes with the same SuperMicro server. -A. ___ 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] Passing CallerID to SIP phone from TDM400P
Do you have a wait(2) before your dial(SIP/) ? You need to allow a full ring before you build your first sip packet. Jeremy Bogan wrote: Yeah I have callerid=asreceived in my zapata.conf still nothing unfortunately. I get that when the calling party has caller id blocked on their end. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Should ZAP channels pass CNAM to SIP?
signalling=pri_cpe callerid=asreceived I see that I get the callerID CNAM in the cdr records, but the same information does not show up on the display on my Cisco 7960 phone only the ANI. I do get Callerid from voip to voip calls . Just not on the zap to voip calls. My question is does anyone have CNAM passing through to voip? I am trying to figure out if I have a configuration error or this is a system limitation. Sure would be nice to see the name of the weirdos that call me. ___ 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] CNAM callerid from a T100p to sip cisco 7960 not working.
I have callerid setup on my PRI coming into my T100p and I know this works because I can see the CNAM in my cdr records. But even with callerid=asreceived set in zapata.conf I only get ANI to the sip devices. Any ideals what I could have goofed up? [channels] rxgain=2.0 txgain=2.0 echocancel=yes hidecallerid=no usecallerid=yes restrictcid=no callwaitingcallerid=yes echotraining=yes context=default switchtype=national signalling=pri_cpe callerid=asreceived group=1 channel=1-23 ___ 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] Cisco 7940/7960 and voicemailmain not able to press keys after a hold.
I have noticed a problem with the Cisco 7940/7960 phones where if you put your voice-mail box on hold using soft keys and come back you can no longer navigate. I am curious if anyone else can duplicate this problem. Happens reliably for me with the 7940 phones. I use rfc2833 for DTMF. I would think it was a Cisco bug, but for the fact that this did not happen with older version of Asterisk. ___ 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] ex-girlfriend logic not working in latest CVS?
Ex-girlfriend logic not working in latest CVS? Incoming sip calls don't work. Anyone else seen this problem? Extension logic looks good: exten = 6153248305/_931NXXX,1,Queue(queue1); exten = 6153248305/_615NXXX,1,Queue(queue2); ;exten = 6153248305,1,Queue(queue3); show dialplan looks good: -- Added extension '6153248305' priority 1 (CID match '_931NXXX')to vantage -- Added extension '6153248305' priority 1 (CID match '_615NXXX')to vantage ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR - Asterisk integration
I would be interested. Tenorio, Leandro wrote: Seshu, I'm interested could u provide more info... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu Sent: Wednesday, July 14, 2004 11:02 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SMDR/CDR - Asterisk integration Hi All, The CDR Tool in .PHP is working great. We have put this into production. Here is the Link: http://67.109.153.236/asterisk-stat/cdr.php If anyone is interested, I will generously contribute the code for your use. Seshu Kanuri ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Cisco 7960 problems persist (for me, anyway)
I checked-out CVS Head today to get realm support, I have over hundred Cisco phone on my servers and I have not noticed any Qos problems. You may want to check the duplex of your switches and Asterisk boxes. If you don't have full duplex, that is more then likely your problem. Brian Cuthie wrote: It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box using a TDM400P interface). Cheers, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sip and Digest realm
Olle E. Johansson wrote: James Sizemore wrote: Has anyone else changed the Digest realm? Did you have any odd problems? In the chan_sip2 module, I've a setting called realm= in sip.conf Time to port that over to chan_sip. No, it doesn't cause any harm. On the contrary, the RFC states that this should be unique for your service. The password belongs to the realm, not really the from: domain or your server's name. One server could handle multiple realms and one user could actually have to authenticate to multiple realms for one call. Thanks, I see that the patch went into cvs today, I'll pull it out a give it a test (Not that there is much chance of this patch messing any-e-thing up.) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Call transfer with RTP transfer as well?
Make sure you have canreinvite=yes in all peers in sip.conf that the call goes through. Also making sure that you don't have tT on any of your Dial statements in extension.conf. But your real problem is that you have some type of network problem use mii-tool eth0 at a bash prompt, and make sure you are full duplex on both boxes as well as on the switch. You should be able to have dozens of call chaining through Asterisk boxes with out voice quality problem, even on very modest hardware. Robert Bedell wrote: I am using SER as a proxy, and using Asterisk as a PBX. A user calls in to a 1-800 number. They listen to the IVR on one Asterisk PBX, and then are transferred to the call center at the other Asterisk PBX. Calls are being brought into the system via SIP. I need to transfer users from one Asterisk box to the other. Functionally this works fine, practically it doesnt as Asterisk forces the RTP stream to go through the first box into the second. That kills latency and makes the calls unusable. Has anyone else had a similar problem? Ive been looking for a while, and am now fairly experienced with Asterisk. Is there a way I dont know of to get Asterisk to do the SIP call transfer? Is there a way I can signal back to the SER proxy not to hang up the call but to transfer it if I cant get Asterisk do what I want without hacking it? Im perfectly capable of adding this functionality to Asterisk if necessary, I just dont want to spend the time if there is already a way to do this. Maybe Im doing something stupid and dont realize it. Thanks! Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk -- Cisco router
Check the duplex on your ethernet conection on both the Cisco and the Asterisk box. Make sure neither are half duplex. Joseph wrote: What codec should be used to connect a * box to a cisco router which has a t1 with 24 trunks coming in? My router voip dial plan looks like this: dial-peer voice 2 voip destination-pattern [1,2,,3,5,8].. session protocol sipv2 session target ipv4:10.x.x.x dtmf-relay cisco-rtp codec g711ulaw no vad ! The problem I have is when more than one call is on it, sometimes the quality gets very bad. If more than one access the conference room it starts to blip real badly. Thots, ideas greatly appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
Some firewalls when doing nat will alter the return address (need to make nat work) but not recalculate the header checksum, (Sonic walls come to mind.), Linux will proply delete any tcp/udp packet that fails its checksum at the kernel level, and send an error to the app. If this is happening to you Asterisk should log some kind of error. AstGrp wrote: Update... I did some more testing today.. And with the same setup but one box behind a Linksys router and another box behind a Pix firewall.. The linksys works with no problems... So it appears to be how the PIX is handling NAT SIP... If any one has any thoughts on this , it would be greatly appreciated. And thank you James for the support you have given today. Thanks, gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of AstGrp Posted At: Friday, March 12, 2004 4:29 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Do I need to associate the outside interface of the PIX with the phone on the inside.. I don't remember doing this before... Setup * Server --- PIX FW --- WWW CLOUD PIX FW --- IP Phone Again the only difference than before is the First PIX FW Old setup was (Different server though) * Server Linksys Router WWW CLOUD PIX FW IP Phone Any thoughts? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Friday, March 12, 2004 2:58 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call The pings are pinging the out side port on the nat device, You don't have a rule in your nat table to associate it with a device on the inside. You should reset the phone and then see if the qualify shows a return time. You will need to make the phone register every time you change you config till the qualify shows a time. A good way to do this is to reboot the phone. Your nat device will have a default time that it keep nat rules in its table. Your qualify time will need to be lower then this value. AstGrp wrote: ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
Make sure your using qualify=500 in the sip.conf along with nat=yes, make sure any firewalls allow 5060 udp and tcp and random ports above 1 in form your PBX. If you have all that it should work. AstGrp wrote: Yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Thursday, March 11, 2004 10:47 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call You do have : nat_enable: 1 nat_received_processing: 1 On the Ciscos? AstGrp wrote: I am having a similar problem... I get the same message, but inbound calls can go through This is only Cisco phones that are behind NAT. I have tried your recommendations from below, but still no luck.. User can make outbound calls, just can't receive any. Any ideas would be greatly appreciated.. I even tried to change the timeout value in chan_sip, but it just waits longer to fail.. Just dosen't seem to want to communicate... Thanks, gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Posted At: Tuesday, March 02, 2004 11:46 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Are you using Cisco phones. ? I had this issue with my cisco phones. I didn't had any issues with dropped calls. All I did to fix this was set a prefered_codex and set proxy_register to 0. I hope this helps. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call *CLI Mar 3 12:55:05 WARNING[1150495040]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) This has been brought up in the previous post but it does not seem to have an answer for it so far. I cvs the stable v1.0 this morning after compiling and installing I have calls drop 1 minutes into the connection with the above message. If anyone has any idea of this occurrence. I have set up sip.conf: canreinvite=no -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.
Andrew Gillham wrote: Sounds good. I have not been that bothered with it when I make a normal voice call. It is mostly annoying when hitting the messages button on the phone. My delay helped that situation. Perhaps on calls where asterisk is proxying the rtp stream we could have an option to tell asterisk to open the connection to the 7960 before the connection is setup on the other side of the call. So the 7960 gets a head start. It would force the codec but that is fine by me, my G.729 is preferred and I don't mind asterisk transcoding since I have a low number of calls. -Andrew I think Barton found the root problem, Native bridging fails or takes to long to setup causeing the delay. I am going to see if a bug has already been opened on this and if not do so. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
I have noticed that sometimes you need to comment out profiles with nat=yes on and then reload, then uncomment them and reload, for Asterisk to clean out historical settings. Try that. I have run phones before on odd port with out trouble, so I don't think that is your problem. AstGrp wrote: Ok.. Let me start by saying that SJPhone works fine through NAT and the Cisco phones inside the internal network work fine also... It's just the Cisco phones on the outside using NAT. For Testing I opened the Firewall open on the IP for the * Server. I have done, everything you recommended below, but still no go... When the phone registers with port 2842? Not the standard 5060? Any ideas? I believe this is where my problem sits... Thanks, -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Friday, March 12, 2004 9:03 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Make sure your using qualify=500 in the sip.conf along with nat=yes, make sure any firewalls allow 5060 udp and tcp and random ports above 1 in form your PBX. If you have all that it should work. AstGrp wrote: Yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Thursday, March 11, 2004 10:47 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call You do have : nat_enable: 1 nat_received_processing: 1 On the Ciscos? AstGrp wrote: I am having a similar problem... I get the same message, but inbound calls can go through This is only Cisco phones that are behind NAT. I have tried your recommendations from below, but still no luck.. User can make outbound calls, just can't receive any. Any ideas would be greatly appreciated.. I even tried to change the timeout value in chan_sip, but it just waits longer to fail.. Just dosen't seem to want to communicate... Thanks, gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Posted At: Tuesday, March 02, 2004 11:46 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Are you using Cisco phones. ? I had this issue with my cisco phones. I didn't had any issues with dropped calls. All I did to fix this was set a prefered_codex and set proxy_register to 0. I hope this helps. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call *CLI Mar 3 12:55:05 WARNING[1150495040]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) This has been brought up in the previous post but it does not seem to have an answer for it so far. I cvs the stable v1.0 this morning after compiling and installing I have calls drop 1 minutes into the connection with the above message. If anyone has any idea of this occurrence. I have set up sip.conf: canreinvite=no -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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
Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
The pings are pinging the out side port on the nat device, You don't have a rule in your nat table to associate it with a device on the inside. You should reset the phone and then see if the qualify shows a return time. You will need to make the phone register every time you change you config till the qualify shows a time. A good way to do this is to reboot the phone. Your nat device will have a default time that it keep nat rules in its table. Your qualify time will need to be lower then this value. AstGrp wrote: Ok... If put in the qualify=500... It says it is unreachable... But ping times Are fine... PING 69.133.182.77 (69.133.182.77) from 10.100.254.21 : 56(84) bytes of data. 64 bytes from 69.133.182.77: icmp_seq=1 ttl=241 time=54.9 ms 64 bytes from 69.133.182.77: icmp_seq=2 ttl=241 time=52.0 ms 64 bytes from 69.133.182.77: icmp_seq=3 ttl=241 time=54.2 ms 64 bytes from 69.133.182.77: icmp_seq=5 ttl=241 time=57.9 ms 64 bytes from 69.133.182.77: icmp_seq=6 ttl=241 time=56.0 ms 64 bytes from 69.133.182.77: icmp_seq=7 ttl=241 time=54.0 ms Any thoughts there? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Friday, March 12, 2004 11:50 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call I have noticed that sometimes you need to comment out profiles with nat=yes on and then reload, then uncomment them and reload, for Asterisk to clean out historical settings. Try that. I have run phones before on odd port with out trouble, so I don't think that is your problem. AstGrp wrote: Ok.. Let me start by saying that SJPhone works fine through NAT and the Cisco phones inside the internal network work fine also... It's just the Cisco phones on the outside using NAT. For Testing I opened the Firewall open on the IP for the * Server. I have done, everything you recommended below, but still no go... When the phone registers with port 2842? Not the standard 5060? Any ideas? I believe this is where my problem sits... Thanks, -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Friday, March 12, 2004 9:03 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Make sure your using qualify=500 in the sip.conf along with nat=yes, make sure any firewalls allow 5060 udp and tcp and random ports above 1 in form your PBX. If you have all that it should work. AstGrp wrote: Yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Thursday, March 11, 2004 10:47 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call You do have : nat_enable: 1 nat_received_processing: 1 On the Ciscos? AstGrp wrote: I am having a similar problem... I get the same message, but inbound calls can go through This is only Cisco phones that are behind NAT. I have tried your recommendations from below, but still no luck.. User can make outbound calls, just can't receive any. Any ideas would be greatly appreciated.. I even tried to change the timeout value in chan_sip, but it just waits longer to fail.. Just dosen't seem to want to communicate... Thanks, gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Posted At: Tuesday, March 02, 2004 11:46 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Are you using Cisco phones. ? I had this issue with my cisco phones. I didn't had any issues with dropped calls. All I did to fix this was set a prefered_codex and set proxy_register to 0. I hope this helps. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call *CLI Mar 3 12:55:05 WARNING[1150495040]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) This has been brought up in the previous post
Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.
exten = 6500,1,Answer exten = 6500,2,Wait,1 exten = 6500,3,VoicemailMain2 Or should I say, Me too! Is this the bug for the case in question? CSCed48311: Media takes 0.4 sec to be set up Thanks. -Andrew Yes the problem is that when making outgoing calls, there is enough of a delay in the call setup once the remote side picks up, that people that answer the phone hello will be heard saying o or if they talk fast enough not heard at all therefor leaving a very awkward silence at the start of a call. This is very annoying. A earlier person suggested answering the calls before dialing and playing a ringing sound till the start of the voice. That may be a work around of sorts for some, you will hear a ring then a congestion tone on call that can't connect, or a ring before a operator messages (say to dial one before the number) that most users may not be used to. I'll be playing with that ideal to see what odd effect a ring has before call setup causes. The work around may be less annoying then the problem. smile I'll see. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
You do have : nat_enable: 1 nat_received_processing: 1 On the Ciscos? AstGrp wrote: I am having a similar problem... I get the same message, but inbound calls can go through This is only Cisco phones that are behind NAT. I have tried your recommendations from below, but still no luck.. User can make outbound calls, just can't receive any. Any ideas would be greatly appreciated.. I even tried to change the timeout value in chan_sip, but it just waits longer to fail.. Just dosen't seem to want to communicate... Thanks, gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Posted At: Tuesday, March 02, 2004 11:46 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Are you using Cisco phones. ? I had this issue with my cisco phones. I didn't had any issues with dropped calls. All I did to fix this was set a prefered_codex and set proxy_register to 0. I hope this helps. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call *CLI Mar 3 12:55:05 WARNING[1150495040]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) This has been brought up in the previous post but it does not seem to have an answer for it so far. I cvs the stable v1.0 this morning after compiling and installing I have calls drop 1 minutes into the connection with the above message. If anyone has any idea of this occurrence. I have set up sip.conf: canreinvite=no -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 and short delay before voice star ts after ring.
Thanks for the information. You have saved me a few hours on the phone with TAC. smile Low, Adam wrote: We have a TAC case open on this issue (reference DDTS CSCed48311) as well, apparently it was going to be fixed in a 7.1 release (yes I know we're only at 6.2 but that's what Cisco stated) but now we are hearing that it will not be fixed in that release but would most likely be further down the track. The issue is specific to SIP on 79xx phones, H.323/SCCP/MGCP all work fine. We're hoping to get a *special* release of the bug fixed SIP code for testing within the next 3/4 weeks. If we get it I'll post an update ... -Original Message- From: Duane [mailto:[EMAIL PROTECTED] Sent: 03 March 2004 15:12 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring. Bisker, Scott (7805) wrote: I think what James is referring to is the delay once the call already been dialed. It's not specific to Ciscos, as I'm experiencing the same problem on my polycom phones. Must be SIP related. The problem is that once a call is dialed, when the remote party picks up the phone, the first half second is cutoff. The remote party won't hear the first half second of the call. I had this happend several times in the last few days. I've also had a few complaints from users recently. Here's what it looks like. I noticed the same issue using a SIP soft phone, I can't recall having the same issue with a IAX soft phone, pretty sure it didn't happen... I'm testing now to see if I can make it happen, but it seems to be fine... ___ 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] Cisco 7960 and short delay before voice starts after ring.
When calling out on a Cisco 7960 there is a short delay before the call gets setup and the other side can hear your voice. Anyone know how to compensate for this effect? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls always parked on 701
I can't believe you would add anymore digits to listen for. I have thought about speeding up the digit play back. It seems to take forever when waiting for 7.0.1 smile Jim Sneeringer wrote: Actually, it works fine as long as the parkpos values are numbers. If you put in a * or #, it seems to ignore what you supply and start with 701. I just happened to be starting with a *. -Original Message- From: Jim Sneeringer To: [EMAIL PROTECTED] Date: Wed, 25 Feb 2004 13:48:47 -0600 Subject: [Asterisk-Users] Calls always parked on 701 Reply-To: [EMAIL PROTECTED] No matter what I put in parking.conf for parkpos, I find that the first call is always parked on 701. Is this a bug? Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream transfer into outer space
You could always create a rule to match any-e-thing 3 or 4 digits, that always forwards to the receptionist [match_all_local] exten = _NXXX,1,Goto(receptionist|s|1) exten = _NXX,1,Goto(receptionist|s|1) [trunk] include = localnumbers include = match_all_local include = international include = longdistance [default] include = trunk Jim Rosenberg wrote: The Grandstream BudgeTone 101 phone has a Transfer button. This appears to be a blind transfer: once you've dialed the extension to which you want to transfer, the phone tries to do this and then dumps you out. My question is this: Let's say I explain to my users that I don't want them using the Transfer button, to use # and let Asterisk transfer the call, or to use parking, and again let Asterisk handle it. But, someone forgets. They hit the Transfer button anyway. Then they type the wrong extension. If they had transferred using the # key and let Asterisk do it, Asterisk would have reacted reasonably to a wrong extension, but the Grandstream doesn't know about all this magic. So: now I've got my caller just sitting there, transferred into nowhere. Is there a way to pick the caller up? I haven't found a way to do this. When this happens the caller is still connected to something, and at the Asterisk console, sip show channels shows the call. It seems as though there ought to be some way to reach in and connect to it ... Any ideas welcome. These Grandstream phones are kind of nice. I sure don't want to have to start out a new installation by *taping over* the Transfer button, but if there isn't a way to reach a stranded caller, it's deadly. -T.i.A., Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA 186 Registration!!!!
You can only use the r option if you answer the call first exten = 106,1,Answer exten = 106,1,Dial(SIP/106,30,tr) other wise remove the r Erick Weber V. wrote: Thank you very much I just make the change and I'm up an running. One more quick question, why I can not hear the ring in the phone connected to the ATA, My extensions are configure as follow: exten = 106,1,Dial(SIP/106,30,tr) Thanks for the quick response Best Regards Erick - Original Message - From: Ejay Hire [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, February 23, 2004 7:01 PM Subject: RE: [Asterisk-Users] ATA 186 Registration Hi. Open http://ip.of.your.ata/dev and set LoginID0: and LoginID1: to your login Id. Then set UseLoginID: to 1. If you haven't already, set a password on the ATA by entering it into the UIPassword: field. This field does not have repeat to confirm, so type carefully. Last but not least, there is a bug in the v3.0 code for the 186's. If you use the TOS bit's to mark SIP for QOS, downgrade back to 2.1.6. In 3.0, the ata sets TOS to 0x0, and ignores the TOS: configuration field. Hope that helps, Ejay Hire ISDN-Net Network Engineer ...Providing VoIP services to Tennessee businesses since 2003 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Weber V. Sent: Monday, February 23, 2004 5:24 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ATA 186 Registration I'm tring to register my ATA to * and I getting the following message: Feb 23 18:13:04 NOTICE[1125329600]: chan_sip.c:5405 handle_request: Registration from 'sip:[EMAIL PROTECTED] user=phone' failed for 'xxx.xxx.xxx.xxx' I don't know what's wrong an why it register as user=phone??? Coul some one help me Thanks Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] alert-info and Cisco 7960 phones (6.1)
exten = 555,1,SetVar(Bellcore-dr1) Will do what you want. Andreas Anderson wrote: Hiya, Getting the 7960 to use the Bellcore-dr1 through dr5 was fairly straightforward. The release notes indicate that you can trigger other ringtones on the phone (in the section Support for SIP Alert-Info Header), but I can't get anywhere with it. the only thing i'm getting, is using exten = 555,1,SetVar(ALERT_INFO=7) to get the phone to ring Ring Ring... Ring Ring with the preset Ring Type. It doesn't matter what alert_info i set, no matter if i use Bellcore-dr1 or anything else. What are you using, and how does the phone react to it...? regards, aa _ Find your perfect match @ http://personals.xtramsn.co.nz with XtraMSN Personals! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] alert-info and Cisco 7960 phones (6.1)
This will give you what you want.I type a little to fast for the brain buffer sometimes. exten = 555,1,SetVar(ALERT_INFO=Bellcore-dr1) Andreas Anderson wrote: Hiya, Getting the 7960 to use the Bellcore-dr1 through dr5 was fairly straightforward. The release notes indicate that you can trigger other ringtones on the phone (in the section Support for SIP Alert-Info Header), but I can't get anywhere with it. the only thing i'm getting, is using exten = 555,1,SetVar(ALERT_INFO=7) to get the phone to ring Ring Ring... Ring Ring with the preset Ring Type. It doesn't matter what alert_info i set, no matter if i use Bellcore-dr1 or anything else. What are you using, and how does the phone react to it...? regards, aa _ Find your perfect match @ http://personals.xtramsn.co.nz with XtraMSN Personals! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec matching weirdness
A better option and one Asterisk desperately needs is some kind of --lint option, Which would check the config for errors and useless misspelled options. smile I personal find one or more typos or misspelling a month, On my PBXs. Eric Wieling wrote: Maybe someone will write a patch to print an error to the console if reinvite= is found in the config file.? On Sat, 2004-01-17 at 15:44, Olle E. Johansson wrote: Dustin Goodwin wrote: I did find something interesting. If you set reinvite=yes then * can setup the RTP stream so that it avoids the media proxy in the * box completely. I haven't tested to see if it changes anything. Can we please kill reinvite - it does not exist in the SIP channel as an option for anything. Period. There is an option called canreinvite that you can set to yes or no. Setting reinvite to anything will not change anything at all. However, setting canreinvite to something will change ASterisk's behaviour during a SIP call. It may also break your conversation if your SIP device does not support the SIP re-invite mechanism. Please read: http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+canreinvite for more information. /Olle PS. I know that the reinvite option is mentioned in many archived e-mails, which does not help at all. Please do not add any more messages with this option, as it will only confuse users. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GrandStream Budgetone Phone DHCP General Observations
I'm interested. TeleSIP wrote: I'll try to hack a NAT friendly tftp server on monday. Are you still looking for it? I found one if you need it. Let me know and I will post the info. Andres. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] Turning up the volume on outgoing sip to sip gateway calls?
I have a need to make the outgoing volume on a meetme room louder for an all sip setup. I could use txgain if I had Zaptel devices in the loop. I do not think I have that option with sip? Any Ideals? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk freezing HELP
Do you type reload at the cli a few times a day? If so try not reloading Asterisk and I'll bet Asterisk stop blocking. If you don't normally reload the box you will need to trouble shot normally. mattf wrote: Hello, I have had several instances over the last month of Asterisk freezing, sometimes after 12 hours, sometimes after 8 days. The common elements are that: - all Zap channels lock[hangups don't register and no new calls in or out] - no new in/outbound calls can be made on Zap or SIP channels - people who are still connected to calls can continue to talk - in the CLI interface, you can show channels and view other info like sip show peers, but you cannot do a stop now only way to stop asterisk process is a kill -9 - all attempted connections to the manager interface time out - the processor load and RAM usage on the machine is low. I am experiencing these freezes on two separate identical RedHat 9 machines. On each I have 4 T1s connected and a usual concurrent call volume of 45 Zap/SIP conversations at once 14 hours a day. The processor load is never that high and the RAM usage is less than half. There are never any consistent errors or warnings in the logs or the CLI. I do have several AGI perl scripts in the dialplan and the SIP phones that connect are almost all Grandstream 102's, That's all I can think of that may be causing any problems. I have mpg123/musiconhold deactivated. Has this happened to anyone else out there? Does anyone have any suggestions? or ideas as to what may be causing it? Thanks for any help, MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Uptime counters (was RE: [Asterisk-Users] Bayonne and Asterisk)
You can not rotate logs with out dropping calls, and if logs get a little over 2Gbs Asterisk will crashes... So I could figure out the average time between crashs just by log level and call volume! LOL This is with out running into a single bug. smile Thankful I can restart Asterisk from time to time myself, but for a person that can not go down this would be a sticking point that would need fixing! Steven Critchfield wrote: On Tue, 2003-11-18 at 09:53, Florian Overkamp wrote: At 09:45 18-11-2003 -0500, you wrote: And yes, they can run fine together(I'm not using VOIP, just a T1 out of Asterisk to Bayonne to test and see if it would work). The IVR application that I currently still have running on Bayonne is only still on Bayonne because it can never go down, and Bayonne has proven itself to me to be extremely stable, while I cannot personally say AT THIS TIME that an Asterisk box would stay up for over 6 months with no crashes. Actually in this light it might be cute to have an 'uptime' counter inside asterisk (maybe a lastlog that can also show the reason of the last restart - was it a stop gracefully or did it just crash?) *grin* Hmm, maybe it wouldn't be much of a hack to get at the show uptime information and dump it with each log as it is sent to the events.log file so you can see if certain length runtimes cause crashes as well. Especially with respect to the post I just read about the user who has asterisk going nuts every day. I wouldn't be opposed to it being put in the CLI prompt too, or maybe just made available and then we could do something like the PS1 formatting of the prompt. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Uptime counters (was RE: [Asterisk-Users] Bayonne and Asterisk)
What bug # should I look for you patch under? smile Andrew Kohlsmith wrote: You can not rotate logs with out dropping calls, and if logs get a little over 2Gbs Asterisk will crashes... Why not? Why are the logfiles kept open for the entire life of Asterisk? Hell even my heavily loaded qmail server isn't this braindead in that regard. If * can be patched to open, write, close for every log write it is trivial to rotate logs: mv /path/to/logfile /path/to/logfile.old while fuser -s /path/to/logfile.old ; do sleep 1 ; done bzip2 -1 /path/to/logfile.old and you're done. mv does not change the inode, so asterisk does not notice it if it _is_ in the middle of a write, and the fuser do/while loop waits patiently until asterisk is done with the file. Next time Asterisk tries to open the file it will fail (since it doesn't exist) and will recreate it. Piece of cake. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Uptime counters (was RE: [Asterisk-Users] Bayonne and Asterisk)
I did not even know about it! But seeing as it is not in the change log no wonder? You have the bug number the notes are under for usage? [EMAIL PROTECTED] asterisk]# grep log ChangeLog -- Agent Callback-login support -- Added Dialogic VOX file format support Andrew Kohlsmith wrote: You can not rotate logs with out dropping calls, and if logs get a little over 2Gbs Asterisk will crashes... Why not? Why are the logfiles kept open for the entire life of Asterisk? Hell even my heavily loaded qmail server isn't this braindead in that regard. Maybe I missed part of this thread, but as of like 10/05/03 cvs there was a new app added for this called (I think) logrotate. It's supposed to allow you to send * a remote command and rotate your logs. I upgraded for this feature but have not had time to test it yet, it's on my look at list. Like I said maybe I missed part of thread but you should be able to setup a cron job and forget about it. Anybody using the logrotate app? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hunt groups and SIP?
I would like to setup a hunt group, not a group ring, using sip phones. Anyone done this with sip devices? Comments suggestions? I have not had much luck with the outgoinglimit=1, incominglimit=1 stuff that I would need to get busy extinctions to work right, which is why I'm asking on the list. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF
Vocal has redundancy. Asterisk has features. They both have bugs. smile What a choice! costas wrote: I can't resist asking. What do you think of Vocal as compared to *? Anything Vocal has but missing in *? -- Original Message -- From: Scott England [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Mon, 17 Nov 2003 02:58:55 -0800 I am trying to connect to a vocal server from an asterisk server. A call is received via iax2 to my asterisk server. I then initiate a SIP connection to the vocal server. everything works great except dtmf doesnt work. A cisco 5300 can connect to this vocal server and do dtmf without a problem. I have my dtmf set to rfc2833 in the general section of the sip.conf . I can confirm that the channel is in rfc2833 during the call via show channel. With SIP debug though I dont see any event for dtmf. I do see dtmf in IAX though. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP calls no longer work
My guess is you need something like this: disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm allow=ilbc allow=speex allow=lpc10 Andrew Thompson wrote: - Original Message - From: jerk face [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 17, 2003 3:53 PM Subject: [Asterisk-Users] SIP calls no longer work Hello, I'm having a problem with SIP. More specifically, I can't make any calls using SIP. I have had an iConnectHere account and Free World Dialup account working for quite some time, and now all of a sudden I can't make any SIP outgoing calls. PBX*CLI sip show registry Host Username Refresh State 192.246.69.223:5060 X120 Registered 213.137.73.178:5060 120 Registered 213.137.73.178:5060 120 Registered The above CLI output shows that I am registered to FWD and my iConnectHere accounts. The following output shows what I see when I make a call using FWD or iConnectHere: Starting simple switch on 'Zap/4-1' -- Executing SetCallerID(Zap/4-1, X) in new stack -- Executing SetCIDName(Zap/4-1, XX) in new stack -- Executing Dial(Zap/4-1, SIP/[EMAIL PROTECTED]) in new stack == Everyone is busy at this time -- Hungup 'Zap/4-1' Running 'sip debug' does not solve this problem. Running 'sip debug' is to allow you to determine the problem, or capture the info necessary for someone onlist to determine the problem. Turning it on doesn't solve problems by itself(not on purpose, anyway). Now that you've turned it on, attempt a call, and paste in the messages for us. I'm out of ideas, has this happened to anybody before? Thank you for your time. __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users - Andrew Thompson Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise.^+$Rf)+-^+$RXb+rXb+r+-w-z rs== ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Radius on *
Amen Lars Boegild Thomsen wrote: I won't agree that RADIUS shouldn't ever have been deployed in a VoIP environment. While it can be argued that RADIUS is not in any way an ideal solution and it can also be argued if it is necessary in a PBX software such as Asterisk, fact is that many IP service providers already have advanced billing systems based on RADIUS in place. As I see it RADIUS does not add another layer of complexity - but instead another layer of abstraction. If Asterisk could do authentication and accounting using RADIUS it wouldn't need to bother with different RDBMS's and the whole discussion of PostgreSQL versus MySQL would be void :) Freeradius easily support MS-SQL, MySQL, Oracle, Sybase, PostgreSQL etc. etc. without any problems whatsoever, so it would significantly simplify things if Asterisk had a smart RADIUS module. Actually even the provisioning of dial-plans could be done through RADIUS, which would make mass-deployment of Asterisk boxes a lot simpler to administrate. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax over SIP alaw/ulaw
You will need to check with Cisco to see if the ATA188 has the same issues with faxing as the ATA186. http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_field_notice09186a0080094af7.shtml Dave Weis wrote: Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. Thanks dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP and Goto failures?
I have been having a lot of problems with SIP calls and
gotos within contexts as well as between contexts.
They work some of the time and fail some of the time
but the console reads the same either way. Am I the
only one having this problem? A little sample config
below.
[macro-stdexten]
exten = s,1,Dial(SIP/${ARG1}|20|t)
exten = s,2,Voicemail2([EMAIL PROTECTED])
exten = s,102,Voicemail2([EMAIL PROTECTED])
[isdnnet]
exten = 61,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = ,1,Goto(61|1)
exten = 8500,1,VoicemailMain2([EMAIL PROTECTED])
___
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Re: [Asterisk-Users] Clearing Queue Stats?
Just a note, there is a bug with queue stats when you reload Asterisk dynamic users will not reset to 0 calls but as Troy suggested default queue members will. Meaning that if you stay in your set and take calls all day and the guy next to you removes himself from the queue to go smoke, when he logs back in his queue answered calls will not reset when reload is run and yours would. With some ring strategies this may really bit. smile I use ringall for all my queues so this does not effect me. If this effect you you may want to open a bug ticket on it. Another thing I noticed with queues that may be a bug, is that when a dynamic user logs out and then logs back in he loses his number of answered calls. I guess the real fix would be to make it so all queue members dynamic and default never lose there number of answered calls even if they log out and then back in from the same number. Also a reload really should not clear the queue answered calls but a new command clear queue should. smile By the way none of this queue clearing stuff has anything to do with Troys patch which works really well! big smile David C. Troy wrote: If you do an asterisk -rx reload from a cron job daily it will clear the stats nondestructively. You might check out my queue patch which adds several features (queue position/holdtime announcement, timeout) as well as abandoned/completed call counters: http://bugs.digium.com/bug_view_page.php?bug_id=214 http://bugs.digium.com/file_download.php?file_id=304type=bug Dave = David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales ToadNet - Want to go fast?410-544-1329 FAX 570 Ritchie Highway, Severna Park, MD 21146-2925 www.toad.net On Sun, 2 Nov 2003, Ken Godee wrote: Is there a way to clear the Queue stats? That is with out restarting *? I'd like to reset them daily and don't see a way to do that. Unless the only way is maybe a cron restart asterisk like every weekday @ 04:00? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP protocol bug ???
Does this have a bug number so I can track it? Jan Janak wrote: On 07-11 13:17, John Todd wrote: From what I can understand of the issue you describe, it sounds like the problem resides on the remote side, and not Asterisk's side. You are sending an invalid request in your first query, and the remote side is sending Unauthorized, meaning that it believes you have supplied credentials, but they are the wrong credentials. This is the end of the conversation, since both sides have given their final words on the subject. Unauthorized means that the message contained no credentials or the server was unable to verify the credentials. When a user agent (asterisk) gets an Unauthorized message then it is supposed to retry with proper credentials. What arguably _should_ be happening is that the remote SIP host should be sending 407 Proxy Authentication Required, but it's not. Therefore, Asterisk is behaving correctly. This is not a bug in Asterisk. That depends on the type of the remote host. Registrars, PSTN gateways, and user agents send 401, proxies send 407. In any case asterisk should be able to handle both for any type of message except ACK and CANCEL (which can not be challenged). Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM to Run VoIP On Linux
Asterisk would need scalability and redundancy on the voip side to play in the soft-switch area. The biggest issue stopping Asterisk having redundancy and scalability using sip is the inability to work with just about any sip device without canreinvite turn off. If Asterisk could handled reinvites correctly you could setup fallback and/or redundant gateways to the PSTN network. Making it a shoe in for large installs. As it is Asterisk just can not scale from a Voip perspective. SER would need to have some kind of PSTN trans-coding. But it can scale! Vocal has the redundancy and scalability, but no real PSTN trans-coding. also Vocal also has serious quality control issues. So of the big three free, yeah Asterisk would be a good place to start. Although Vocal on paper is a little better though out. Asterisk has a lot more working features. But I would bet money IBM uses none of the above. smile Mark Spencer wrote: Asterisk has got to be about the best kept secret in telephony. I've seen numerous articles on slashdot about VoIP, even in relation to Linux and only *once* has the post even mentioned Asterisk. Am I missing something, or is Asterisk clearly a good potential player in any kind of linux-based soft-switch idea? Mark On Sat, 8 Nov 2003, Dave Cotton wrote: For those who don't wake up at 5.00 am and start reading /. http://searchnetworking.techtarget.com/originalContent/0,289142,sid7_gci935769,00.html -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IBM to Run VoIP On Linux
Besides you got list four times since May!smile http://slashdot.org/search.pl?query=asterisk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstreams can't call out with latest CVS
Thanks a bunch you were on the money. Do you know about when that changed? John Todd wrote: Grandstreams phones can't call out with the latest CVS, anyone know what the last good CVS date was? You may be experiencing difficulty due to bad codec permissions, since the latest CVS updated version (earlier today) does seem to work for me. Make sure you have correct allow/disallow permissions. Experiment with per-peer and then [general] configs for both. Some relevant settings from sip.conf (note: these are not optimal and should not be considered valid for all users. They simply work at this 5-minute snapshot of time for me. I change these settings on an almost hourly basis): [general] disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm allow=ilbc allow=speex allow=lpc10 ; my grandstream 102 [2209] type=friend username=2209 secret=nosecretpasswordhere host=dynamic context=internal canreinvite=yes nat=1 dtmfmode=info qualify=100 disallow=all allow=ulaw allow=alaw JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA186 configuration for fax application
The ATA186 is only rated to 9600 baud, it is not usable for faxing as most faxs are 19200. See Cisco sight for details. http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_field_notice09186a0080094af7.shtml Eric Wieling wrote: If you have ANY chance of sending a fax over VoIP your codec MUST be ulaw or alaw. On Wed, 2003-10-29 at 03:52, Manuel Marín García wrote: I have problems to send faxes using a fax machine connected to a ATA186 line 2. My sip.conf is [1151] type=friend username=1151 secret= canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=1151 nat=1 context = optica ATA 186 connection mode is 0x00460400 and Audio mode is 0x00150014 I always get poor line condition in the fax machine Does anyone have an example of how to configure ATA186 and sip.conf Please help? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstreams can't call out with latest CVS
Grandstreams phones can't call out with the latest CVS, anyone know what the last good CVS date was? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk ???
To be a true ip tel softswitch, Asterisk would need SS7 support. No one is working on SS7 signaling for Asterisk. WipeOut wrote: Victor Medrano wrote: Asterisk will become a real ip tel softswitch or is going to other way ? like vovida regards It is already an IP softswitch.. Or may be I don't understand you question.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call pickup (*8) on SIP devices.
Yes Ing. Angel Gomez Garcia wrote: Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
10 Fix call waiting tone. 9Fix the tftp configs so that I can host my own provisioning server. Or make a command prompt based tool kit, so that I can use Gaps with out writing a http screen scraper. 4 Having the Conference button do something would be cool. John Brown (CV) wrote: Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things like ring tones and fixing call waiting are already on the list. :) Lets also keep the replys away from gripes and complaints and more towards constructive comments. I'll be taking the results and sending GS a summary. John Brown, Chagres Technologies, Inc Buy your VoIP hardware from us email: sales at chagres d0t net for quotes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
Agreed, don't drive up my shipping cost. light is good. Tilghman Lesher wrote: I'd have to respectfully disagree. If this is really a problem I'd suggest taking advantage of the mounting bracket on the bottom and either attach the phone to the desk or attach a sheet of lead. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] Proper quoting (was: NAT, SIP
learn.to can kiss my *** smile I'll top quote till death, And so will almost every other person on this planet. grin Tilghman Lesher wrote: On Tuesday 14 October 2003 18:15, Uriel Carrasquilla wrote: I have to tell you, at the expense of offending you, that I use MS-Outlook and the responses go to the tope of the messages. At work I use Lotus Notes and the same thing happens. Before, I used PROFS (on mainframes) and the same principle applied. All in all, 20+ years of using this principle for e-mails at both work and home. As a matter of fact, I am of the opinion that the response to E-mails should go at the top to save time. However, this is not about me but the * group and the well being of this list. Does anybody else have a strong opinion one way or the other? If it is left to John and myself we have a 1:1 vote. This is all you really need to know: http://learn.to/quote/ -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT, SIP
I wish you would take this stuff to personal email, I am tired of wasting my time reading this crap. If you idiots want to give lesions on how YOU would like people to post on list servers _DO_IT_VIA_PERSONAL_EMAIL_!!! None of the rest of us care. This is a personal messages from you to someone else. Stop wastering MY bandwidth. I personal like top posting as I don't have to scroll all the way to the bottom to read what is most of the time one damn sentence. So, it's not worth *your* time organizing your e-mail sensibly, but it's worth everyone else's time having to dig through lines of text to work out what the context is? I find that selfish, at best. Please see the following page (strong words warning). It pretty much sums it all up nicely: - http://thegestalt.org/simon/quoterant.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bare-bone config
You will need : extensions.conf indications.conf logger.conf manager.conf rtp.conf sip.conf modules.conf ; with a crap load of stuff turned off: noload = chan_modem.so noload = chan_modem_aopen.so noload = chan_modem_bestdata.so noload = chan_modem_i4l.so noload = chan_phone.so noload = chan_mgcp.so noload = chan_iax2.so noload = chan_oss.so noload = chan_iax.so noload = chan_alsa.so noload = chan_oss.so .etc. You may want: voicemail.conf ; Do you want voicemail ? parking.conf ; Do you want to park ? meetme.conf ; Do you want a conf ? queues.conf ; Do you want queues? Conrad Braun wrote: I am just starting to use asterisk as well as VoIP in general, and it's a bit confusing finding out what goes where... in my eyes it seems to be a lot easier to start with a bare minimum, thereby eliminating as many causes for error as possible. when I feel comfortable, I can always expand on top of it. Also, I haven't found any documentation on which files are read and in what order - are the names hardcoded? why isn't there a h323.conf? so it's also a matter of curiosity I guess ;) WipeOut wrote: Why do you want to remove some of the conf files? Just leave them all there.. its not like they use up a lot of space or anything.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Parking and Paid Digium software modifications
Most PBX do park the way your old KSU system did. As a matter of fact Asterisk is the only PBX I have ever seen that parks the way it does. If given a choice my uses would use the normal way. And I would be happy not to here the question can you speed up her talking? LOL Andrew Kohlsmith wrote: Is there an underlying reason you want to do this? Because if a call is already parked on 701 and you transfer another call to 701 to park it, both callers would be connected. Actually I have to agree with Matt; I would like to be able to specify where it's parked and get a busy if I try to park a call where there is already one waiting. That's how the old KSU worked anyway. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queues and max time in queue timeout?
Can a call be kicked out of a queue if it reaches a specific timeout? I don't see an obvious way to do this in either queues.conf or extensions.conf any pointers or patches to do this? smile ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RedHat 9.0 and 100 percent CPU utilization
Red Carpet will give you some serious dependency problems later down the road. Bisker, Scott (7805) wrote: I found the best way to upgrade is install Red Carpet from www.ximian.com. Subscribe to the RH 9.0 channel. And do a complete update. The only drawback is that this method doesn't update the kernel. To do the kernel, ftp the latest kernel from updates.redhat.com. rpm -ivh latest kernel.rpm. Change /etc/grub.conf to reflect the newest kernel is the default. The Redhat way way is: up2date --upgrade-to-release=release-versision The easies way is to cruse on over to www.freshrpms.net and get a copy of apt for Redhat and then do a apt-get dist-upgrade release-versision ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gastman and SIP?
I have been testing Gastman and Astman with SIP calls. As I have no Zap phones, so I have a few question on what is normal behavior? When a call comes in and I have created extensions for all phones (example: Channel = SIP\3846) Should the little lines connect between the pre-made extension or should they pop up temporary icons with no connection to the hand made extensions? The Green light does light up. What should Invite and Originate do, right now they just ring a phone once and hangup. Anyone know of any other programs that I can be tested for call status and redirection? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemailmain2 user docs?
Has anyone browsed through the source code and made a list of menu option for VoiceMailMain2? Or know of some user documentation hiding in Internet land some place? If not there well be soon. Ho hum. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Status of shipdate on the 4 port FX0 card?
Does any-e-one know if the 4 port FX0 cards will be shipping anytime soon? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemailmain2 user docs?
I have looked both these URLs over , neither is a User level description of the menu choices. I'm trudging through the code now, the only thing I have found so far that is not listed in the voice mail prompts is that you can press 0 if you have a o extinction in the same contest. I was hoping there were some keys for the users to skip the greetings or a key to goto the VoiceMailMain from VoiceMail. But have not seen any yet. Olle E. Johansson wrote: James Sizemore wrote: Has anyone browsed through the source code and made a list of menu option for VoiceMailMain2? Or know of some user documentation hiding in Internet land some place? If not there well be soon. Ho hum. Start here http://www.voip-info.org/wiki-Asterisk+cmd+VoiceMailMain2 Also check here http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf ...and if you don't find what you're looking for, please help us add more information to the wiki. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration between *'s
Here are a few outgoing gateway configs that work for me. [vocal] type=friend host=1.1.1.7 insecure=1 port=5065 accountcode=memrtr ;dtmfmode=info [cisco] type=friend host=1.1.1.3 insecure=1 canreinvite=no port=5060 dtmfmode=info accountcode=memrtr Xisco wrote: That's true if always there to connect two asterisk servers, but I'm doing some proves in order to connect one asterisk server with another SIP server. That's the matter. - Original Message - From: Jamie Carl [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 19, 2003 12:12 PM Subject: Re: [Asterisk-Users] SIP registration between *'s ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 + SIP
[EMAIL PROTECTED] tftpboot]# cat OS79XX.TXT P0S30100 Get this image as well. Shaun Ewing wrote: Hello all, I know this isn't strictly Asterisk, but I'm sure that there are more people here using the Cisco 7960 w/ SIP, so I thought I'd post here. I've just bought a Cisco 7960 phone to use with Asterisk. It came with the CallManager image on it. I've got the 4.4 SIP images (P0S3-04-4-00). If I put P0S3-04-4-00 in the OS79XX.TXT file, the phone downloads this fine (watching TFTP server debug). It then proceeds to request P0S3-04-.bin. I don't know why. Naturally this file isn't found. I tried renaming the file to P0S3-04-.bin. The phone then downloads around 80% before aborting. I hope somebody might be able to shed some light on the situation. Any help would be greatly appreciated. Thanks, Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7206 as SIP-PSTN Gateway?
I use both Ciscos and Asterisk as Sip gateways to pstn. I can say a lot of good things about both, and a few bad things as well. The Ciscos are a very solid product with very good very fast tech support. It also has some really nifty fax detection with redirection via email options (AS5300 and AS5800). Ciscos can not hire-pin voip calls, This makes it hard to handle congestion between multiple units. Cisco configs syntax for dial plans are ugly as hell, but you can have over lapping dial plans with weights. I have never had a Cisco gateway crash on me that was not running T code. Asterisk can do hire-pining of voip calls (You can send voip calls to another gateway if you have reached max out-bound channels! via voip) Asterisk has a very pretty syntax for dial plans. The problems with Asterisk as a voip gateway using Sip is it can be a bit unstable I have yet to have a week go by that I have not needed to restart Asterisk when it is acting as a voip to pstn gateway. I have written watch dog scripts that restart Asterisk when it dies or locks-up. And support for problem with Sip in a gateway scenario is slow at best. Very few people are using Asterisk as Voip gateways using SIP . So it is very hard for the people involved to reproduce your problems to fix them. I like the price of the hardware I would have bought a large number of T400 cards and used Asterisk if it was more stable doing SIP to PSTN, but alias I just use Asterisk in small town gateways because I can not justify the price of a Cisco for small gateways. I however can not over state the usefulness of hire-pining and when Asterisk gets better at being a voip gateway I will be deploring it in larger instillations. Building IVR with Asterisk is much easer then doing the same on a Cisco you don't need to know tcl. smile All in all if price is your number one factor go with Asterisk if however stability is you number one factor go with a Cisco. More notes: I am currently using 36xx systems as SIP gateways in some locations. There are VoIP NM cards for those platforms, though they are NOT cheap, even on the used market. You'd be much better served from an economic standpoint by getting a slew of el-cheapo rackmount PC's and using Digium cards. Even if the failure rate is higher (which, in my experience, is not the case,) you can do failover quite easily (easier?) using IAX2 to your edge devices. Plus, in my opinion, the Ciscos are lacking many features that Asterisk provides as a gateway device. (Can you compile your own software on your Cisco? and many others which are obvious and on which I will not elaborate.) Cards: NM-1v or NM-2v for POTS NM-HDV-1T1-12 - 1 port, 12 channels NM-HDV-[1,2]T1-[24,48] - 2 ports, 24 or 48 channels There are equivalent E1 cards, change the numbers to match (30 or 60 ports) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Security vulnerability report
If one is using SIP the CVS-current can be extremely unstable. I would say about half the time I have tried a new CVS checkout on a test box. (about once a week) I have had lockups or missing features. I like Asterisk and CVS but with out testing in a semi large environment the cvs -current is anything but stable for me, I am guessing you mostly use ZAP devices? Tilghman Lesher wrote: Odd, I've found CVS-current to be extremely stable, so I run it on all of our production machines. No machine is ever more than a couple weeks out of sync with CVS (except for a few machines in the field which I can't get to right now). -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Callgroup, Pickupgroup and SIP
Know bug http://bugs.digium.com/bug_view_page.php?bug_id=116 Pertti Pikkarainen wrote: I have problems with this as well ( similar config ). My CVS is 10 days old. I can get the call picked up with *8 ( *8# does not work ) but the phone B never stops ringing. B rings forever. I'm using SNOM200. --Pertti WipeOut . wrote: I have just started to play with callgroups and pickupgroups.. I updates my * from CVS this morning (about 15 mins ago).. I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf.. I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB.. Have I not configured somthing correctly or is there a bug?? Later. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Callgroup, Pickupgroup and SIP
know bug http://bugs.digium.com/bug_view_page.php?bug_id=116 WipeOut . wrote: I have just started to play with callgroups and pickupgroups.. I updates my * from CVS this morning (about 15 mins ago).. I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf.. I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB.. Have I not configured somthing correctly or is there a bug?? Later. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] *78 *72 and sip?
I know *8 kind of works with SIP but what about the rest should they work do they work with a zap device? *0# sends flash *8# remote call pickup (pickup phone in your group) *67# disable caller id *70# no call waiting *78# do not disturb on *79# do not disturb off *72# enable call forwarding *73# disable call forwarding *82# enable callerid ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *78 *72 and sip?
I agree the * functions should be selectable and doing this at the dial plan level seems a good place to put them. Having not look at the code. I don't know how hard this will be. Digium wants $150 an hour for contract work, I wonder if Mark could give me a quote on adding these features either to the sip channel drive or to the dial plan? I would need a good ball park figure to get a PO cut. If anyone thinks they can do this work in a timely manner I would not mind a quote from you as well. (Wish I had the time to do it myself. Would be a good way to get my head around this code base.) Brian West wrote: I agree they should stay at the dialplan level. bkw On Mon, 8 Sep 2003, John Todd wrote: I know *8 kind of works with SIP but what about the rest should they work do they work with a zap device? *0# sends flash *8# remote call pickup (pickup phone in your group) *67# disable caller id *70# no call waiting *78# do not disturb on *79# do not disturb off *72# enable call forwarding *73# disable call forwarding *82# enable callerid Personally, I'm not in favor of seeing any of these features find their way into the channels, Zap, SIP, or otherwise. This should all be done at the dialplan level, and not at the invisible level of the channel driver. I created a large list of features, which implement almost all of the standard CLASS featureset, but all the logic is in the dialplan, and not in the phones or channels. Even call pickup and flash should be handled in the dialplan for Zap drivers, through applications, IMHO. See the bug report/feature request on this topic: http://bugs.digium.com/bug_view_page.php?bug_id=071 JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *78 *72 and sip?
How hard do you think it would be? Tilghman Lesher wrote: On Monday 08 September 2003 01:56 pm, Brian West wrote: I agree they should stay at the dialplan level. It's not a matter of staying; it's a matter of moving. Those features are already present within the chan_zap channel driver. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call parking -- what was the key combination?
If you put Tt in your dial statement you can type # some number to transfer to. Of if you can send flash hooks that will work as well. Dave Alan Caruana wrote: what i'm asking is what is the key sequence you have to dial for the transfer .. it was something like *7# if I remember well, I know I had it working, but the client lost the paper I wrote it on for him, and I can't trace the email I got it from! cheers Dave - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 05, 2003 3:11 PM Subject: Re: [Asterisk-Users] call parking -- what was the key combination? To park a call you simply transfer the call into extension 700 (this is the default and can be changed).. To get the call back you just dial the parked location.. If you are using an IP phone this is a problem becasue it will not tell you the location of the parked call so you will not know where to collect it from.. hi great gurus of asterisk :) could somebody remind me the key combination to send a call into the parking queue ? while you're at it, are there any other key combinations I should know?? eg. put a call on hold etc. thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] remotely picked-up extension keeps ringing
Yes its a known issue. http://bugs.digium.com/bug_view_page.php?bug_id=116 Louis-David Mitterrand wrote: Hello, As of today's cvs * snapshot I am able to pickup a ringing (sip) cisco 7960 with *8 but the extension then keeps ringing indefinitely, even though I picked up the call. Is this a known issue? Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STUN server from Vovida
The client device has to support stun. Bugetones do, ATA do, 7960 don't..etc Dave Cotton wrote: On Wed, 2003-09-03 at 09:01, WipeOut . wrote: Sorry to answer a question with a question.. Can stund and * be loaded on the same server and run at the same time? I've also never been able to figure out stund, if that is possible wouldn't it be the answer to most of the SIP difficulties. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT - Headsets for Cisco 7940/7960
Someone told you wrong. Works fine, volume is a little low however without powered headsets. Erik Anderson wrote: I converted my Cisco 7960 to SIP. I did not try the headset port because I was told that Cisco did not enable the headset port for SIP. Erik -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of DUSTIN WILDES Sent: Wednesday, September 03, 2003 8:02 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] OT - Headsets for Cisco 7940/7960 This is Off-Topic for Asterisk, but I wanted to get some feedback on headsets for Cisco 7940/7960 phones. We have about 10-20 people who wants/needs a headset for their phone was hoping to collect some real-world input. Thanks!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is the DTMF bug in bugs.digium.com what number.
Yes this is the bug, please update the ticket with what you said in email and confirm the bug. http://bugs.digium.com/bug_view_page.php?bug_id=171 Brenton D. Rothchild wrote: Hi all, I'm not sure if this is the same thing, but if so, let me know and I'll submit more info to bugs.digium.com. I'm using an AudioCodes MP-104 FXO box, with snom 200 handsets: snom200 --- asterisk --- MP104FXO --- POTS When using rfc2833 or inband (using G.711u/a), either way (verified with 'sip show channel ...'), the DTMF tones come across as very short - and in the case of '1' and '9', they are very screechy. If this is something anybody wants to see more about, I'll be happy to provide more info. Thanks, Brenton Rothchild - Original Message - From: James Sizemore [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 25, 2003 1:42 PM Subject: Re: [Asterisk-Users] Is the DTMF bug in bugs.digium.com what number. I have entered a bug report under bug_id=171 If you are also using sip and having this problem please confirm this bug and enter any comment or note you have on this bug. James Sizemore wrote: Is the DTMF duration bug in bugs.digium.com what is the number? I did not see a bug that reflected this issue, But I only looked through the first few pages. This is becoming a major bug, Just need to know if I need to enter a report. Again this is the bug that when using any DTMF tones they are so short that remote IVRs do not detect them. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF tones not long enough on out going calls
DTMF tones are not long enough on out going calls, when I'm using either info or rfc2833. Does anyone know if the tone length value is in rtp.c or chan_sip.c ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgetone Defective Units
I have not had any problem at all with the 10 I have. They sound good and work well. The only problem I ever had was a problem with remote ntp servers. Andres wrote: Hi, I would like to know if others have experienced a high percentage of Budgetone defective units. We purchased 4 to test with our Asterisk. One was DOA and the other died after 3 days. So far we have a 50% failure. This does not look good. Let me know if its just us or if it is widespread. On the other hand we have purchased about 50 ATA186s and none of them have failed. Thanks, Andres ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Transfer
Blind and assisted transfer work with Cisco 7960 phones. Blind transfer works fine with Budgetones. As long as you register to Asterisk. Jamie Carl wrote: Ok, just been thinking about this and thought I would ask before trying it out again. What is the state of SIP transfers? By this I mean transfers initiated via SIP messages, not via DTMF and '#'. Last time I tried, on X-Lite, clicking the transfer button dropped the call. Also, are/will both REFER and BYE/also methods be supported? To me, the SIP way of transfering is alot nicer and it seems silly to me to have a transfer button on your SIP phone that u can't use. Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] '#' doesn't work for me
Do you have transfer turn on in zapata.conf? transfer=yes Hi, I cannot use '#' to initiate transfers. I have tried on different phones (7960, ATA, X-Lite). When I press '#' during a call, nothing happen. I have both T and t switches in Dial application. The transfer function works with Flash key on ATA, but in a very strange wayThe final destination is hunged up and then automatically called by the initial caller... This behavior request to put on hook the phone connected to the ATA in order to accept the transfer. During this period the phone is busy for the caller, so I must use some tricks in the dialing macro in order to acomodate this. Any other suggestions to better solve the transfer function? BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Park and out-going trunk calls.
If you add t to you out-going trunk Dial lines: exten = _NXX,1,Dial(SIP/[EMAIL PROTECTED]||t) exten = _NXX,2,Congestion so that you can still use park to park a call or transfer the phones, You have a problem of not being able to use # on external IVR systems. Is there any solution to this problem? Other then training hundreds of users not to try and park calls that you originate. smile ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
