[Asterisk-Users] Asterisk E1 and Cisco as5300
I am trying to send calls from an AS5300 to Asterisk via e1 and I get this bit of information in place of routing information Going to extension s|1 because of Complete received Accepting call from '' to 's' on channel 1, span 1 Here are the relevant zaptel and zapata pieces. span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 signalling=pri_cpe switchtype=national context=pritest group=1 channel = 1-15,17-31 Any help will be greatly appreciated. Thank you, Christian Hoffmeyer YottaDot Solutions Huntsville, AL (iax) 700.859.4508 ___ 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] ISDN WAN ISDN bridge possible?
Patrick, doe a google search for ISDN over IP, maybe that's your solution. jo Patrick Stuckenberger wrote: Hi list, is it possible to create something like a ISDN-WAN-WAN-ISDN bridge? We have to change our location, but our number and the telephone system should shoulb stay the same. kind regards, Patrick Stuckenberger ___ 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] Czech sound files
Hi, if there is somebody working on Czech support please contact me off list, so we can work together. Petr Mosnicka -- YieldTech - linuxova divize ATAX Group, spol. s r.o. V zavetri 6 tel: +420-777-2LINUX 170 00 Praha 7 mailto: [EMAIL PROTECTED] Ceska republika http://www.YieldTech.cz ___ 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] New ENUM service, what do you think?
While I wish the guys at Stealth the best of luck, I'll say again that ENUM is _NOT_ the solution for VoIP routing in the current real world. See the mailing list archives for more of my rants on why DNS is not the answer for cost-based routing (where cost is monetary, distance, qos, or any other comparative metric.) TRIP (RFC 3219) is the answer, but I'm the only one pounding that drum, it seems. If anyone here on the list has $100,000 to put together a real programming effort towards getting that implemented, y'all let me know. The longer this waits, the more lame and broken become the solutions offered. sigh JT At 1:28 PM -0400 5/2/04, Joe Baptista wrote: On Sat, 1 May 2004, Dean Collins wrote: Yes but no information about how this will operate, what regulation or restrictions on joining, what connection protocols will be used etc etc agreed - you see alot of business fluff - but the technicals are very important and on many of these ventures they fail to include them. regards joe www.baptista.god Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Reid A. Forrest Sent: Saturday, 1 May 2004 8:21 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] New ENUM service, what do you think? From http://www.thevpf.com/ To join, please e-mail [EMAIL PROTECTED] or telephone 1-212-232-2020 (Mon-Fri 9AM-5PM EST). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jimfl Sent: Saturday, May 01, 2004 5:11 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New ENUM service, what do you think? Jim/frank, Can you give us more information about how to access this enum? I've been to the stealth web site and there is no information about access. I look forward with interest to what you have up and running today for asterisk users to benefit from. Cheers, Dean Sorry, I am not associated with Stealth in any way. Just saw the news story and thought it would be of interest to Asterisk users. It sounds like you don't have to be a VOIP provider to get access to their service. They talk about businesses using the service. If anyone finds out how to get access to their service, please post. 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] Probs with oh323 driver: audio only in 1 direction
Hi,
try to setup asterisk as an ISDN2H323-Gateway. The only problem
i have after establishing a call is, that Audio works only from IP to
ISDN-Phone but not from ISDN to IP-Phone.
A known problem ???
Thanks in advance
Michael
i am using asterisk-cvs, pwlib V1.6.6 (janus), openh323 V1.13.5 (janus)
and oh323-0.6.0
Here are my config's
##
# modem.conf #
##
[interfaces]
context=isdn
driver=i4l
language=en
stripmsd=0
dialtype=tone
mode=immediate
msn=8540340
context=8540340
device = /dev/ttyI0
device = /dev/ttyI1
##
#extensions.conf #
##
[general]
static=yes
writeprotect=no
[8540340]
exten = s,1,Wait,1
exten = s,2,Answer
exten = s,3,Dial(OH323/192.168.70.227)
[voip-h323]
exten = s,1,Answer
exten = s,2,Dial,Modem/ttyI1:${OH323_DSTE164}
##
# oh323.conf #
##
[general]
listenAddress=192.168.70.1
listenPort=1720
connectPort=1720
tcpStart=1
tcpEnd=2
udpStart=1
udpEnd=2
fastStart=no
h245Tunnelling=no
h245inSetup=no
inBandDTMF=no
silenceSuppression=no
jitterMin=20
jitterMax=100
ipTos=none
outboundMax=10
inboundMax=10
simultaneousMax=10
wrapLibTraceLevel=1
libTraceLevel=1
libTraceFile=stdout
gatekeeper=192.168.70.1
gatekeeperTTL=600
userInputMode=TONE
amaFlags=default
accountCode=H323
context=voip-h323
[register]
alias=isdn
gwprefix=8
[codecs]
codec=G711A
frames=20
--
Michael Niehren __ _ powered by
/ / (_)__ __ __
/ /__/ / _ \/ // /\ \/ /
//_/_//_/\_,_/ /_/\_\
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Re: [Asterisk-Users] Resolved: sipgate.de
I know it's exciting to get things working, however,
there are some things wrong with your configuration, despite it perhaps
working ok.
Is it really? You can make outbound calls this way?
In your friends definition (friend-sipgate) you don't have a host
specified.
host=sipgate.de
Without that I doubt you can make any calls, since asterisk won't know where
to
send the call to.
Further, since you're using fromdomain, it should be the authentication
realm,
which is sipgate.de, not sipgate.net. But this won't hurt your call
completion
Fromdomain will get placed into the From headers instead of your ipaddress.
some domains are picky about it when you're using special services and they
want to make sure you're actually a domain member.
Also, your localnet= parameter should be the network address, not the host
address, but you're probably ok, since the mask cuts it off.
Since you don't have a valid friends definition, your incoming calls come
into
the default context, and you need to be carefull what you make available
there.
It's never a good idea to have calls coming in this way, without restriction
or authentication.
Enjoy
- Original Message -
From: Jay Milk [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, May 03, 2004 15:23
Subject: [Asterisk-Users] Resolved: sipgate.de
(History: Getting my home asterisk system set up; one land-line,
multiple SIP trunks; goal is to have a wife-proof transparent phone
system)
Just wanted to let everyone know that I got sipgate.de working with my
asterisk system; relevant settings below:
--account--
Sipgate number 8001234 (change to suit yours)
Password password
--network--
External static IP routes to internal 192.168.254.204 (static)
--sip.conf--
[general]
port=5060
bindaddr=192.168.254.204
externip=x.x.x.x ; insert your external IP here
localnet=192.168.254.204
localmask=255.255.255.0
nat=yes
register = 8001234:[EMAIL PROTECTED]/99049 ; 99049 =
incoming/Germany
[friend-sipgate]
username=8001234
secret=password
fromuser=8001234
fromdomain=sipgate.net
type=friend
nat=yes
dtmfmode=rfc2833
canreinvite=no
--extensions.conf--
exten = _01149.,1,Dial(SIP/0${EXTEN:[EMAIL PROTECTED],30)
exten = _01149.,2,Congestion
exten = 99049,1,Wait,1
exten = 99049,2,Answer
exten = 99049,3,Dial(SIP/sipura2b,30)
Hope this helps someone else. The register allowed me to receive
incoming calls, but outgoing calls failed until I set the externip and
nat settings.
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Re: [Asterisk-Users] New ENUM service, what do you think?
You may be quite right, I have read parts of the rfc at least, I remember, but the lure of using cheap existing infrastructure is probably to great. KHB - Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, May 04, 2004 03:20 Subject: RE: [Asterisk-Users] New ENUM service, what do you think? While I wish the guys at Stealth the best of luck, I'll say again that ENUM is _NOT_ the solution for VoIP routing in the current real world. See the mailing list archives for more of my rants on why DNS is not the answer for cost-based routing (where cost is monetary, distance, qos, or any other comparative metric.) TRIP (RFC 3219) is the answer, but I'm the only one pounding that drum, it seems. If anyone here on the list has $100,000 to put together a real programming effort towards getting that implemented, y'all let me know. The longer this waits, the more lame and broken become the solutions offered. sigh JT ___ 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] Asterisk - no outband DTMF with Mediatrix
Dear List members, I have this problem with Mediatrix 24-FXS-line gateway and out-of-band DTMF. It seems not working - the RTP mode is not working and when I select INFO mode, the Mediatrix is behaving just the same as with RTP mode. Here is a bunch of logs to explain this: 1. The RTP out-of-band mode (dtmfmode=rfc2833): This is OK reply from Asterisk to Mediatrix when RTP mode selected. Seems OK ;-): [...] SIP/2.0 200 OK CSeq: 1091919829 INVITE v=0 o=root 35059 35059 IN IP4 xxx s=session c=IN IP4 xxx t=0 0 m=audio 12210 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 [...] And then, during connection with asterisk, when we use DTMF, this shows on debug: [...] May 3 17:49:42 NOTICE[139648000]: rtp.c:418 ast_rtp_read: Unknown RTP codec 96 received May 3 17:49:42 NOTICE[139648000]: rtp.c:418 ast_rtp_read: Unknown RTP codec 96 received [...] 2. INFO mode (dtmfmode=info): Proper INVITE from Mediatrix: [...] INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 CSeq: 1657017135 INVITE Content-Type: application/sdp Contact: Port 3 sip:[EMAIL PROTECTED] Supported: replaces User-Agent: MxSipApp/4.4.11.68 MxSF/v3.2.7.30 v=0 o=MxSIP 4563726510189014186 6429835688411497953 IN IP4 xxx s=- c=IN IP4 xxx t=0 0 a=sendrecv m=audio 5004 RTP/AVP 8 18 4 0 13 111 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:111 X-nt-inforeq/8000 [...] And then nothing happens, Asterisk shows no DTMF events. Thanks for any help, Arek Bekiersz ___ 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, park and conference
Hi! I have 2 grandstream budgetone 100 series. I can call allright, but I cant do call transfer, park and call conference. (all features works with tdm devices) the 1. Check if Asterisk is always in the media path, i.e. you need the t or T option (or something similar) in your Dial statement. Alternatively you could introduce a canreinvite=no in sip.conf for the GS phones. 2. Check your context setup in extensions.conf and make sure that in call cases your GS phone has the parkedcalls context available Philipp ___ 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] Siemens cordless phone
The SDK for the Siemens USB cordless phone was just released a few days ago. I understand from a few people I spoke with when this was first released that this could be ported to work for Asterisk. Does anybody have any thoughts now they have seen the sdk information? Cheers, Dean Gigaset M34 USB PC Adapter offers an open interface to enable third party solution providers to integrate cordless phones into their applications for VoIP, messaging and home control. To encourage developers Siemens provides a free Software Development Kit (SDK), Internet-based support, as well as 24 hours hotline via the Siemens mobile developer portal: http://www.siemens-mobile.com/developer. The portal contains detailed information about the interface, the SDK and the hardware and tool environment. image001.gif
Re: [Asterisk-Users] Asterisk remains in the media path
Le lun 03/05/2004 à 18:48, Jeremy McNamara a écrit : Actually its cuz chan_h323 sucks like that. Correct me if I'm wrong, but I browsed the archives and I got the feeling that you (Jeremy) were one of the main developers of the chan_h323... aren't you a little harsh about your own work? :-) Anyway, is there any plan in the chan_h323 roadmap to support direct RTP between endpoints? Thanks, Paul ___ 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] New ENUM service, what do you think?
John Todd wrote: TRIP (RFC 3219) is the answer, but I'm the only one pounding that drum, it seems. If anyone here on the list has $100,000 to put together a real programming effort towards getting that implemented, y'all let me know. The longer this waits, the more lame and broken become the solutions offered. sigh One small oversight in your thinking, something like TRIP will only benefit large telcos and VOIP providers with interconnects, I don't see this flowing down to a tangible benefit to the average person, where as something like enum.164 is. TRIP is based on BGP and BGP already does most of the IP routing smarts TRIP is supposed to be beneficial for, however that $100k would be better spent on improving the smarts in the call routing software rather then turning things back into a hub and spoke model, p2p is way more efficient if it can be utilised to it's full potential. At this stage the only potential method to prevent VOIP spam is something like SPF records, which would only end up duplicate enum. It's a lot harder to get phone numbers then IP addresses, so this would overcome people's concerns about dynamically allocated IPs, phone numbers aren't. -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers ___ 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] MGCP: Current CVS works for you?
Hi there, I have serious problems with MGCP and Swissvoice ip10s, and it appears that recent CVS also introduced trouble for other MGCP users. Please check and add comments in the bugtracker so that we can get a clearer picture - thanks! Also comment if things are working fine for you. http://bugs.digium.com/bug_view_page.php?bug_id=0001542 http://bugs.digium.com/bug_view_page.php?bug_id=881 and other MGCP related bugs/fixed. Cheers, Philipp ___ 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] Security Issue in Asterisk with sip.conf configuration.
uhm, strange but does this work on your setup? even with permit and
deny, if a user is not matched in the conf, it is allowed access to the
default context stated in the conf.
On Wed, 2004-04-28 at 16:12, James H. Thompson wrote:
I think the problem is that using permit= alone does nothing.
You need to combine it with a deny= as in:
deny=0.0.0.0/0.0.0.0 ; deny all
permit=123.123.123.123 ; allow only this address - netmask defaults to:
/255.255.255.255
order matters, the deny needs to come first.
for reference here is the code from acl.c that checks the rules:
int ast_apply_ha(struct ast_ha *ha, struct sockaddr_in *sin)
{
/* Start optimistic */
int res = AST_SENSE_ALLOW;
while(ha) {
/* For each rule, if this address and the netmask = the net address
apply the current rule */
if ((sin-sin_addr.s_addr ha-netmask.s_addr) == (ha-netaddr.s_addr)
res = ha-sense;
ha = ha-next;
}
return res;
}
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: William Zhang [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, April 27, 2004 2:43 PM
Subject: [Asterisk-Users] Security Issue in Asterisk with sip.conf configuration.
I had tried many ways with some advanced user help, but without
success(at one point I thought I had it worked).
Here Asterisk is working as a SIP PSTN Gateway, and in the sip.conf
file, there are a lot of entries with just host=a.b.c.d, thinking
that * will only accept calls from host a.b.c.d, but in my test, no
mater how you set up the sip.conf entries, either * will NOT accept
calls for that user account at all, or it will accept calls from any
where without VERIFYING the source IP(whether it is a.b.c.d or not),
so long the sip userid is the username in sip.conf. This post a very
serious security problem.
Of course we can put secret= for each entries, but giving Asterisk GW
and SIP proxy are in 2 TRUSTED IPs, no Authentication is neccessary,
otherwise it increase the SIP traffic quite a bit.
Following are the 4 different entries that I had tried:
#Notice that in the general section, context is pointed to a none
existant context INVALID.
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 212.213.66.68
context = INVALID ;
;srvlookup = yes; Enable SRV lookups on outbound calls
;pedantic = yes ; Enable slow, pedantic checking for
Pingtel
;tos=lowdelay
;tos=184
;maxexpirey=3600; Max length of incoming registration
we allow
;defaultexpirey=120 ; Default length of incoming/outoing
registration
;notifymimetype=text/plain ; Allow overriding of mime type in
NOTIFY
;videosupport=yes ; Turn on support for SIP video
disallow=all; Disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=g729
allow=ilbc
;
;dtmfmode=info
;dtmfmode=inband
dtmfmode=rfc2833
[20034]
type=friend
callerid=TEST 61331045
host=212.213.65.66
nat=yes; This phone may be natted
canreinvite=no
[20035]
type=peers
callerid=TEST 61331045
host=212.213.65.66
nat=yes; This phone may be natted
canreinvite=no
[20036]
type=friend
context=default
callerid=TEST 61331045
host=212.213.65.66
permit=212.213.65.66
nat=yes; This phone may be natted
canreinvite=no
[20037]
type=peers
context=default
callerid=TEST 61331045
permit=212.213.65.66
nat=yes; This phone may be natted
canreinvite=no
Thank you in advance.
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Re: Caller ID Re: [Asterisk-Users] Re: Support Digium
On Sunday 02 May 2004 08:07 pm, Kevin Walsh wrote: Someone on IRC once pointed out the conflict between suggested and must on a similar page and said that their TDM400P (FXS-only at the time) was working on a PCI 2.1 system. Can anyone confirm whether a PCI 2.2 bus is mandatory? Yes, PCI 2.2 _is mandatory_. I know because I just upgraded my TDM400P card on an RMA, and the gentlemen tech emphasized that the motherboard for my new TDM400P card MUST have PCI 2.2. Anon ___ 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] TDM400P FXO, 2 slots?
On Sunday 02 May 2004 08:56 pm, Jamin W. Collins wrote: On Sun, May 02, 2004 at 09:07:37PM +0100, Kevin Walsh wrote: The same Digium shop page suggests that two PCI slots would be required. I'll assume the card is too fat, with the daughter board(s) fitted, to fit into a single slot. This is something I would like to see confirmed, does this card really take 2 pci slots? No, the TDM400P does not take up 2 slots. In fact, in my box I an IDE card in slot 3, next to it I have the TDM400P in slot 4, and a X100P right next to it in slot 5. No space problem at all. Anon ___ 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] Voicemail for Toshiba dk280
On Wednesday 28 April 2004 03:37 pm, Barton Hodges wrote: I would like to use Asterisk for voicemail, connected to a Toshiba dk280. Has anyone done this with this model or similar system? Are there any documents available that could give me some insight into how I can do this? You may want to see: http://www.voip-info.org/tiki-index.php?page=Asterisk%20legacy%20integration ___ 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] How to implement configure agents
Hi I am new to this forum can some body tell me how can i configure and implement agents. if there is any document available on agents implementation plz forward me that thanx Salman __ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover ___ 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] How does Norvergence do it ?
So a guy shows up at the the office, after making an appointment with the office manager / receptionist to talk 'phone systems'. After her eyes glaze over, with him talking T1 and Frame-Relay I get to see him. He's from Norvergence. Well dressed. Tells me they can do a T1 for $79, with unlimited local long distance for free. It also does 'internet'. 'Just give me copies of your phone bill'. I ask some questions, like number porting, like provisioning of DID numbers, like CIR on the data etc. Now HIS eyes glaze over. That's technical talk ... He's just there to follow up on the appointment and 'qualify' the customer to see if we are worthy of their cheap service. After I looked at their website, I can hear 'quack quack'. Cheers, WW Willy Wouters ypOne Publishing ___ 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] How does Norvergence do it ?
Found this: http://w3.ripoffreport.com/reports/ripoff89155.htm Many other nasty stories about them too. -- Cheers, Neil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: 04 May 2004 12:51 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] How does Norvergence do it ? So a guy shows up at the the office, after making an appointment with the office manager / receptionist to talk 'phone systems'. After her eyes glaze over, with him talking T1 and Frame-Relay I get to see him. He's from Norvergence. Well dressed. Tells me they can do a T1 for $79, with unlimited local long distance for free. It also does 'internet'. 'Just give me copies of your phone bill'. I ask some questions, like number porting, like provisioning of DID numbers, like CIR on the data etc. Now HIS eyes glaze over. That's technical talk ... He's just there to follow up on the appointment and 'qualify' the customer to see if we are worthy of their cheap service. After I looked at their website, I can hear 'quack quack'. Cheers, WW Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Timeout Gives T in cdr.
Tilghman Lesher wrote:
On Monday 03 May 2004 13:56, Frank Mandarino wrote:
I have worked around this issue by storing the extension in a
variable, then restoring it using a Goto in the 'T' processing.
For example:
exten = 411,1,SetVar(ORIG_EXTEN=${EXTEN})
exten =
411,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],40,rS(10)) ...
exten = 411,200,Playback(call-timed-out)
exten = 411,201,Hangup
exten = T,1,Goto(${ORIG_EXTEN},200)
I'm curious about your usage here. You don't appear to be using
AbsoluteTimeout, yet you're using extension T, not extension t.
How is this working for you?
This is just an example, not actual working code. It probably should
have used t, but the original message specified T.
In any case, saving the extension in a variable, then restoring it with
a Goto back to the saved extension is the only way I have found to have
the original extension stored in the CDR instead of the somewhat useless
lettered extension.
../fam
--
Frank A. Mandarino [EMAIL PROTECTED]
Spindrift Management, Toronto
416 642-3404
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[Asterisk-Users] Asterisk and windows h.323 gatekeeper calling problems...
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi there, i have a working Microsoft ISA firewall with buildin H.323
Gatekeeper
So Far, i got registerd the asterisk on the M$ Gatekeeper...
here is the h.323 configuration:
; Open H.323 driver configuration
;
[general]
port = 1720
bindaddr = 0.0.0.0
allow=all ; turns on all installed codecs
dtmfmode=rfc2833
gatekeeper = 62.225.189.250
AllowGKRouted = yes
context=local
;
[time]
type=h323
e164=18102341212
context=local
;
;[det-gw]
;type=h323
;prefix=1248,1313
;context=detroit
;
[202]
type=user
host=*
context=incoming
incominglimit=4
Here is the extensions.conf:
debian:/etc/asterisk# cat extensions.conf
[general]
static=yes
writeprotect=no
;
; The Globals category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for
Environmental variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
;DEFAULT EXTENTION
AIC=4455505
REACEND=4455506
AHECHT=4455507
MAILBOX=994801
AIC_MAILBOX=201
AHECHT_MAILBOX=203
REACEND_MAILBOX=202
;SIP EXTENTION
REACEND_SIP=SIP/202
AHECHT_SIP=SIP/203
AIC_SIP=SIP/201
;
;
[local]
include = voice
include = hold
include = meeting
include = demo
exten =
${REACEND},1,Macro(stdexten,${REACEND},${REACEND_SIP},${REACEND_MAILBOX})
exten =
${AHECHT},1,Macro(stdexten,${AHECHT},${AHECHT_SIP},${AHECHT_MAILBOX})
exten = ${AIC},1,Macro(stdexten,${AIC},${AIC_SIP},${AIC_MAILBOX})
exten = ${MAILBOX},1,VoiceMailMain();
exten = time,1,Answer
exten = time,2,Playback,current-time
[voice]
;Voicemail System
exen = 999,1,Voicemail2
[meeting]
exten = 8600,1,Meetme
[hold]
exten = 6600,1,WaitMusicOnHold()
[demo]
exten = 500,1,Playback(demo-abouttotry); Let them know what's going on
exten = 500,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ; Call
the Asterisk demo
exten = 500,3,Playback(demo-nogo) ; Couldn't connect to the demo
site
exten = 500,4,Goto(s,6); Return to the start over
message.
[default]
include = local
[intern]
include = local
[remote]
include = local
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},20)
exten = s,2,Voicemail(u${ARG3})
exten = s,3,Playback(vm-goodbye)
exten = t,1,Dial(${ARG2},20)
exten = t,2,Voicemail(b${ARG3})
exten = t,3,Playback(vm-goodbye)
Now when i want to call time@asterisk-box then it didn't work and I
also get no informations when I tourn on debugging and trace of
h.323... Can somebode give me a configuration of a Gatekeeper gnugk
for example...
Best Regards,
Mark Nicolas
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
iD8DBQFAl4paoKtmDMYNuGsRAsKoAJ98p3OuZecDG719s6I/WywfqUjVxACfdmXj
/fMLqD9BH/p8f3RV8QffLjY=
=tSid
-END PGP SIGNATURE-
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Re: [Asterisk-Users] Beeps clicks and volume problems
On Thursday 29 April 2004 06:20 am, Andres wrote: Sean Garland wrote: I still have problems with beeps and clicks on all my calls. I have polycom sip phones. I also can hear the beeps and clicks on some of my messages, which would lead me to believe that it is more of a decoding problem on the zaptel card. Any ideas? I have seen this happen when the zaptel card is sharing an interrupt with something else, for example a USB bus. You might want to disable unused stuff directly in the BIOS. I too am using a Polycom SIP phone, and noticed the beeps and clicks. I removed a couple of devices sharing the interrupt with the zaptel card, and it helped significantly in reducing the beeps and clicks. Anon ___ 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] Max TE410P card on an Asterisk
Title: Max TE410P card on an Asterisk Hello, Does anybody know the max number of TE410P/TE405P card we can put in an asterisk box? Thanks.
Re: [Asterisk-Users] MGCP: Current CVS works for you?
Hi Philipp, I havn't tried latest mgcp code but I can say that chan_mgcp has serious problems with IP10S that are partially solved by my latest patch http://lists.digium.com/pipermail/asterisk-users/2004-March/041615.html I have received any feedback about it. Regards, Daniel ANDRE Philipp von Klitzing a écrit: Hi there, I have serious problems with MGCP and Swissvoice ip10s, and it appears that recent CVS also introduced trouble for other MGCP users. Please check and add comments in the bugtracker so that we can get a clearer picture - thanks! Also comment if things are working fine for you. http://bugs.digium.com/bug_view_page.php?bug_id=0001542 http://bugs.digium.com/bug_view_page.php?bug_id=881 and other MGCP related bugs/fixed. Cheers, Philipp ___ 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 -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ 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] Max TE410P card on an Asterisk
It depends entirely on the application: number of transcoders running etc. There has been some discussion on this topic in the past - you might consult the archives and the Wiki. Assuming that you're running a fast processor (2.4GHz), I would think the general answer is either one or two 4-port E1 cards maximum in one system. I've had trouble making two 4-port TE410P cards run reliably under heavy load, but I was hammering these channels with a very high rate of call setups in an IVR environment. Regards Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com http://www.evtmedia.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shabanip Sent: Tuesday, May 04, 2004 5:26 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Max TE410P card on an Asterisk Hello, Does anybody know the max number of TE410P/TE405P card we can put in an asterisk box? Thanks. ___ 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] Site for Asterisk-Ethernet Only-Sip Implementation
On Friday 30 April 2004 11:48 am, Akshay Lamba wrote: Hi Everyone, Could someone direct me to a site that talks about Asterisk implementation for Ethernet interfaces/SIP Implementation? I've done my share of googleing and am only able to come up with sites that use digium hardware only. See: http://www.voip-info.org/tiki-index.php http://www.voip-info.org/wiki-SIP Anon ___ 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] Max TE410P card on an Asterisk
Title: Max TE410P card on an Asterisk Short sarcastic answer: (just because I've seen this question 12 times in the last few months!) As many as will fit on your motherboard, but don't expect to use them all :) Long true answer: 2 quad cards on a fast P4 system if you are doing very little VOIP(less than 5 concurrent codec conversion streams). If you are doing a lot of VOIP, then limit yourself to one quad card per machine. MATT--- -Original Message-From: shabanip [mailto:[EMAIL PROTECTED]Sent: Tuesday, May 04, 2004 8:26 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Max TE410P card on an Asterisk Hello, Does anybody know the max number of TE410P/TE405P card we can put in an asterisk box? Thanks.
[Asterisk-Users] Maximum retries exceeded problem...
Searched the
archives thoroughly...
Can't find this
specific problem...
Simple setup with
Asterisk on RedHat. No voice cards in the box, 2 SNOM 200
phones...
Phones seem to work
well, can leave VM, Message Waiting Indicator lights up but when I try to
retrieve messages the call terminates and the following
happens:
-- Executing
VoiceMailMain("SIP/520-a25e", "Mike") in new stack --
Playing 'vm-login' (language 'en')May 4 07:58:07 WARNING[1125329600]:
chan_sip.c:497 retrans_pkt: Maximum retriesexceeded on call [EMAIL PROTECTED]
for seqno 2 (Response)May 4 07:58:07 WARNING[1217602880]:
app_voicemail.c:2748 vm_execmain: Couldn'tread username == Spawn
extension (default, asterisk, 1) exited non-zero on
'SIP/520-a25e'asterisk*CLI
Pertinent section of
extensions.conf
exten = 504,1,Dial,sip/${EXTEN}|10
exten = 504,2,Voicemail(u504)
exten = 504,102,Voicemail(b504)
exten = 504,103,Hangup
exten = 520,1,Dial,sip/${EXTEN}|10
exten = 520,2,Voicemail(u520)
exten = 520,102,Voicemail(b520)
exten = 520,103,Hangup
exten =
asterisk,1,VoicemailMain(${CALLERIDNUM})
Pertinent section of
voicemail.conf
504 = 504,Tech Desk,[EMAIL PROTECTED]
520 = 520,Mike
Picher,[EMAIL PROTECTED]
RE: [Asterisk-Users] How does Norvergence do it ?
Wow, that was GREAT info... I'm in the NY tri-state area so I'm sure I will run into them as competition. Thanks for sending that link - it's unbelievable; also proves the power of such emailing lists where we can share this type of information about unethical companies!Neil Grant [EMAIL PROTECTED] wrote: Found this:http://w3.ripoffreport.com/reports/ripoff89155.htmMany other nasty stories about them too.--Cheers,Neil-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: 04 May 2004 12:51To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] How does Norvergence do it ?So a guy shows up at the the office, after making anappointment with the office manager / receptionist to talk'phone systems'.After her eyes glaze over, with him talking T1 andFrame-Relay I get to see him. He's from Norvergence. Welldressed. Tells me they can do a T1 for $79, with unlimitedlocal long distance for free. It also does 'internet'.'Just give me copies of your phone bill'. I ask somequestions, like number porting, like provisioning of DIDnumbers, like CIR on the data etc. Now HIS eyes glaze over.That's technical talk ... He's just there to follow up onthe appointment and 'qualify' the customer to see if we areworthy of their cheap service. After I looked at theirwebsite, I can hear 'quack quack'.Cheers,WW Willy WoutersypOne Publishing___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo 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-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!?Win a $20,000 Career Makeover at Yahoo! HotJobs
[Asterisk-Users] asterisk + NEC integration
I have an nec electra elite 192 with a t1 card; and am looking for suggestions as to integrating them (can't throw out the system yet!). I have a fully working asterisk server -CVS-04/27/04-19:01:05- (found a hp d220 for $350.00!), 2 digium t100p cards, a plain t1 with loopstart signaling, and 2 working bt102 grandstream ip phones (thanks again Matt for your start from scratch article). This is what I'd like: t1 (loopstart)24 channels | | Asterisk t100p #1 | | Asterisk t100p #2 (best signaling option to NEC - em wink, ls, pri? it will do any of these) | | NEC t1 card - 30 extensions. I found on the wiki David Gomillion's nortel to asterisk (very well done) but he used a pri at both ends. Any help would be greatly appreciated - and I have no problems documenting the process for inclusion to the wiki. t o n y ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Syntax
I've been wondering what the difference is in the syntax of things,
like Dial.
Some examples show things like:
exten = 500,1,Dial,SIP/${EXTEN}|10
but other examples show:
exten = 500,1,Dial(SIP/${EXTEN}|10)
or
exten = 500,1,Dial(SIP/${EXTEN},10)
Which one is correct? Or most correct? Which one is preferred, and why?
I'm sure I'm not the only one with this question... :)
Tim
--
Tim Sailer Coastal Internet, Inc.
Network and Systems Operations PO Box 726
http://www.buoy.comMoriches, NY 11955
[EMAIL PROTECTED] (631) 399-2910 IAX 17003992910
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Re: [Asterisk-Users] Maximum retries exceeded problem...
I don't think your DTMF is set right look in sip.conf for the dtmf
directive for your phones.
cheers!
On Tue, 2004-05-04 at 13:41, Michael Picher wrote:
Searched the archives thoroughly...
Can't find this specific problem...
Simple setup with Asterisk on RedHat. No voice cards in the box, 2
SNOM 200 phones...
Phones seem to work well, can leave VM, Message Waiting Indicator
lights up but when I try to retrieve messages the call terminates and
the following happens:
-- Executing VoiceMailMain(SIP/520-a25e, Mike) in new stack
-- Playing 'vm-login' (language 'en')
May 4 07:58:07 WARNING[1125329600]: chan_sip.c:497 retrans_pkt:
Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno 2
(Response
)
May 4 07:58:07 WARNING[1217602880]: app_voicemail.c:2748 vm_execmain:
Couldn't
read username
== Spawn extension (default, asterisk, 1) exited non-zero on
'SIP/520-a25e'
asterisk*CLI
Pertinent section of extensions.conf
exten = 504,1,Dial,sip/${EXTEN}|10
exten = 504,2,Voicemail(u504)
exten = 504,102,Voicemail(b504)
exten = 504,103,Hangup
exten = 520,1,Dial,sip/${EXTEN}|10
exten = 520,2,Voicemail(u520)
exten = 520,102,Voicemail(b520)
exten = 520,103,Hangup
exten = asterisk,1,VoicemailMain(${CALLERIDNUM})
Pertinent section of voicemail.conf
504 = 504,Tech Desk,[EMAIL PROTECTED]
520 = 520,Mike Picher,[EMAIL PROTECTED]
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RE: [Asterisk-Users] Resolved: sipgate.de
-Original Message- From: Karl Brose Sent: Tuesday, May 04, 2004 2:43 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Resolved: sipgate.de You can make outbound calls this way? In your friends definition (friend-sipgate) you don't have a host specified. host=sipgate.de Without that I doubt you can make any calls, since asterisk won't know where to send the call to. You're correct, I forgot the host= parameter when copying my settings to the list. I assure you it's in my current (WORKING) configuration. Further, since you're using fromdomain, it should be the authentication realm, which is sipgate.de, not sipgate.net. But this won't hurt your call completion Actually, I think that was the difference between it working and not working, if I remember correctly. It was a late night :) Since you don't have a valid friends definition, your incoming calls come into the default context, and you need to be carefull what you make available there. I did not include a context in my example -- fwiw, my full config contains context=german-sip -- I did not think this was necessary to get SIP working for others, and those who know asterisk enough (and many know it better than me, being a newbie of 2 weeks) will likely have their own context settings anyway. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Syntax
On Tue, 2004-05-04 at 08:51, Tim Sailer wrote:
I've been wondering what the difference is in the syntax of things,
like Dial.
Some examples show things like:
exten = 500,1,Dial,SIP/${EXTEN}|10
but other examples show:
exten = 500,1,Dial(SIP/${EXTEN}|10)
or
exten = 500,1,Dial(SIP/${EXTEN},10)
Which one is correct? Or most correct? Which one is preferred, and why?
I'm sure I'm not the only one with this question... :)
They are all correct, but the last one is most like programming and is
preffered by me, and maybe a few others here. The first is historic, and
the second is a mix of the first and third. The second probably should
be avoided as it might break later on if the parser changes.
--
Steven Critchfield [EMAIL PROTECTED]
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RE: [Asterisk-Users] Maximum retries exceeded problem...
Sorry, forgot to include that... Seems to be set right for the Snom phones
(from what I could gather).
[520]
type=friend
secret=blah
host=dynamic
callerid=Mike
dtmfmode=inband ; Choices are inband, rfc2833, or info
defaultip=192.168.0.12
mailbox=520 ; Mailbox for message waiting indicator
;restrictcid=yes; To have the callerid restriced - sent as
ANI
[504]
type=friend
secret=blah
host=dynamic
callerid=TechDesk
dtmfmode=inband ; Choices are inband, rfc2833, or info
defaultip=192.168.0.13
mailbox=504 ; Mailbox for message waiting indicator
;restrictcid=yes; To have the callerid restriced - sent as
ANI
-Original Message-
From: Justin Carlson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 04, 2004 5:00 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Maximum retries exceeded problem...
I don't think your DTMF is set right look in sip.conf for the dtmf directive
for your phones.
cheers!
On Tue, 2004-05-04 at 13:41, Michael Picher wrote:
Searched the archives thoroughly...
Can't find this specific problem...
Simple setup with Asterisk on RedHat. No voice cards in the box, 2
SNOM 200 phones...
Phones seem to work well, can leave VM, Message Waiting Indicator
lights up but when I try to retrieve messages the call terminates and
the following happens:
-- Executing VoiceMailMain(SIP/520-a25e, Mike) in new stack
-- Playing 'vm-login' (language 'en') May 4 07:58:07
WARNING[1125329600]: chan_sip.c:497 retrans_pkt:
Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno 2
(Response
)
May 4 07:58:07 WARNING[1217602880]: app_voicemail.c:2748 vm_execmain:
Couldn't
read username
== Spawn extension (default, asterisk, 1) exited non-zero on
'SIP/520-a25e'
asterisk*CLI
Pertinent section of extensions.conf
exten = 504,1,Dial,sip/${EXTEN}|10
exten = 504,2,Voicemail(u504)
exten = 504,102,Voicemail(b504)
exten = 504,103,Hangup
exten = 520,1,Dial,sip/${EXTEN}|10
exten = 520,2,Voicemail(u520)
exten = 520,102,Voicemail(b520)
exten = 520,103,Hangup
exten = asterisk,1,VoicemailMain(${CALLERIDNUM})
Pertinent section of voicemail.conf
504 = 504,Tech Desk,[EMAIL PROTECTED]
520 = 520,Mike Picher,[EMAIL PROTECTED]
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Re: [Asterisk-Users] Asterisk and windows h.323 gatekeeper calling problems...
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
reacend wrote:
| Hi there, i have a working Microsoft ISA firewall with buildin
| H.323 Gatekeeper So Far, i got registerd the asterisk on the M$
| Gatekeeper...
|
|
| here is the h.323 configuration:
|
| ; Open H.323 driver configuration ; [general] port = 1720 bindaddr
| = 0.0.0.0 allow=all ; turns on all installed codecs
| dtmfmode=rfc2833 gatekeeper = 62.225.189.250 AllowGKRouted = yes
| context=local ; [time] type=h323 e164=18102341212 context=local ;
| ;[det-gw] ;type=h323 ;prefix=1248,1313 ;context=detroit ; [202]
| type=user host=* context=incoming incominglimit=4
|
|
|
|
|
|
| Here is the extensions.conf:
|
|
| debian:/etc/asterisk# cat extensions.conf [general] static=yes
| writeprotect=no ; ; The Globals category contains global
| variables that can be referenced ; in the dialplan with ${VARIABLE}
| or ${ENV(VARIABLE)} for Environmental variable ; ${${VARIABLE}} or
| ${text${VARIABLE}} or any hybrid ; [globals]
|
| ;DEFAULT EXTENTION
|
| AIC=4455505 REACEND=4455506 AHECHT=4455507
|
| MAILBOX=994801
|
|
| AIC_MAILBOX=201 AHECHT_MAILBOX=203 REACEND_MAILBOX=202
|
| ;SIP EXTENTION REACEND_SIP=SIP/202 AHECHT_SIP=SIP/203
| AIC_SIP=SIP/201 ; ;
|
|
| [local]
|
| include = voice include = hold include = meeting include = demo
| exten =
|
${REACEND},1,Macro(stdexten,${REACEND},${REACEND_SIP},${REACEND_MAILBOX})
| exten =
| ${AHECHT},1,Macro(stdexten,${AHECHT},${AHECHT_SIP},${AHECHT_MAILBOX})
| exten = ${AIC},1,Macro(stdexten,${AIC},${AIC_SIP},${AIC_MAILBOX})
|
|
|
| exten = ${MAILBOX},1,VoiceMailMain(); exten = time,1,Answer exten
| = time,2,Playback,current-time
|
|
|
| [voice] ;Voicemail System exen = 999,1,Voicemail2
|
| [meeting] exten = 8600,1,Meetme
|
| [hold] exten = 6600,1,WaitMusicOnHold()
|
|
| [demo] exten = 500,1,Playback(demo-abouttotry); Let them know
| what's going on exten =
| 500,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ; Call the
| Asterisk demo exten = 500,3,Playback(demo-nogo) ; Couldn't
| connect to the demo site exten = 500,4,Goto(s,6);
| Return to the start over message.
|
|
| [default] include = local
|
| [intern] include = local
|
| [remote] include = local
|
|
| [macro-stdexten]; ; ; Standard extension macro: ; ${ARG1} -
| Extension (we could have used ${MACRO_EXTEN} here as well ;
| ${ARG2} - Device(s) to ring ;
|
| exten = s,1,Dial(${ARG2},20) exten =
| s,2,Voicemail(u${ARG3})exten =
| s,3,Playback(vm-goodbye)
|
| exten = t,1,Dial(${ARG2},20) exten =
| t,2,Voicemail(b${ARG3}) exten = t,3,Playback(vm-goodbye)
|
|
| Now when i want to call time@asterisk-box then it didn't work and
| I also get no informations when I tourn on debugging and trace of
| h.323... Can somebode give me a configuration of a Gatekeeper
| gnugk for example...
|
|
|
| Best Regards, Mark Nicolas
|
Append:
works jet... it was a rule problem... So Asterisk works fine with M$
Gatekeeper ;-)
Greetz,
Mark
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RE: [Asterisk-Users] How does Norvergence do it ?
I have a good friend who used to be a sales rep for them. The entire sales pitch is based on making the customer believe that they are lucky to have been offered the opportunity to beome a Norvergence customer as they are extremely selective. If any technical question where to come up, he was trained to let them know that they where not selected to be part of the program and to move on quickly. I do have a copy of their contract if anyone is interested is taking a look at it. It is an interesting piece of legal work. Michael From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J Poz Sent: Tuesday, May 04, 2004 9:45 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] How does Norvergence do it ? Wow, that was GREAT info... I'm in the NY tri-state area so I'm sure I will run into them as competition. Thanks for sending that link - it's unbelievable; also proves the power of such emailing lists where we can share this type of information about unethical companies! Neil Grant [EMAIL PROTECTED] wrote: Found this: http://w3.ripoffreport.com/reports/ripoff89155.htm Many other nasty stories about them too. -- Cheers, Neil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: 04 May 2004 12:51 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] How does Norvergence do it ? So a guy shows up at the the office, after making an appointment with the office manager / receptionist to talk 'phone systems'. After her eyes glaze over, with him talking T1 and Frame-Relay I get to see him. He's from Norvergence. Well dressed. Tells me they can do a T1 for $79, with unlimited local long distance for free. It also does 'internet'. 'Just give me copies of your phone bill'. I ask some questions, like number porting, like provisioning of DID numbers, like CIR on the data etc. Now HIS eyes glaze over. That's technical talk ... He's just there to follow up on the appointment and 'qualify' the customer to see if we are worthy of their cheap service. After I looked at their website, I can hear 'quack quack'. Cheers, WW Willy Wouters ypOne Publishing ___ 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 Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs
Re: [Asterisk-Users] How does Novergence do it ?
My customer is going to ask for a copy of Norvergence's contract to read the details. He said he'd send me a copy when it arrives... From doing a little goggleing it sounds like Norvergence is a scam business model waiting to implode but what do I know. :) Lance ___ 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] Maximum retries exceeded problem...
Also, working this a bit more... if i do the echo
test (extension 600) i get sorta the same thing...
== Spawn extension (default, asterisk, 1) exited non-zero
on 'SIP/520-a25e'May 4 09:15:51 NOTICE[1125329600]: chan_sip.c:5655
handle_request: Unknown SIPcommand 'PUBLISH' from
'192.168.100.12' -- Executing Playback("SIP/520-1a68",
"demo-echotest") in new stack -- Playing 'demo-echotest'
(language 'en')May 4 09:15:58 WARNING[1125329600]: chan_sip.c:497
retrans_pkt: Maximum retriesexceeded on call [EMAIL PROTECTED]
for seqno 2 (Response) == Spawn extension (default, 600, 1) exited
non-zero on 'SIP/520-1a68'
From: Michael Picher [mailto:[EMAIL PROTECTED]
Sent: Tuesday, May 04, 2004 9:41 AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Maximum
retries exceeded problem...
Searched the
archives thoroughly...
Can't find this
specific problem...
Simple setup with
Asterisk on RedHat. No voice cards in the box, 2 SNOM 200
phones...
Phones seem to work
well, can leave VM, Message Waiting Indicator lights up but when I try to
retrieve messages the call terminates and the following
happens:
-- Executing
VoiceMailMain("SIP/520-a25e", "Mike") in new stack --
Playing 'vm-login' (language 'en')May 4 07:58:07 WARNING[1125329600]:
chan_sip.c:497 retrans_pkt: Maximum retriesexceeded on call [EMAIL PROTECTED]
for seqno 2 (Response)May 4 07:58:07 WARNING[1217602880]:
app_voicemail.c:2748 vm_execmain: Couldn'tread username == Spawn
extension (default, asterisk, 1) exited non-zero on
'SIP/520-a25e'asterisk*CLI
Pertinent section of
extensions.conf
exten = 504,1,Dial,sip/${EXTEN}|10
exten = 504,2,Voicemail(u504)
exten = 504,102,Voicemail(b504)
exten = 504,103,Hangup
exten = 520,1,Dial,sip/${EXTEN}|10
exten = 520,2,Voicemail(u520)
exten = 520,102,Voicemail(b520)
exten = 520,103,Hangup
exten =
asterisk,1,VoicemailMain(${CALLERIDNUM})
Pertinent section of
voicemail.conf
504 = 504,Tech Desk,[EMAIL PROTECTED]
520 = 520,Mike
Picher,[EMAIL PROTECTED]
[Asterisk-Users] Help on legacy hardware.
Howdy,
I appologise in advance if this is not the correct forum for this message.
Bought an ACT Networks NetPerformer SDM-9350 Voice/Data router off ebay
('cause it was cheap and has 4 EM/FXO/FXS configurable ports.), now I need
to get it to work.
2 problems:
1) No documentation - I've searched high and low and found very little. Does
anyone have any for this router?
2) I've telnetted into it's console port and it's password protected. Any
ideas? (apart from using it as a boat anchor)
Thanks in advance for any help offerred, I'd very much like to get this
working with *.
Regards,
Stuart.
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RE: [Asterisk-Users] How does Norvergence do it ?
Yes, I would like to see the contract.Michael Miller [EMAIL PROTECTED] wrote: I have a good friend who used to be a sales rep for them. The entire sales pitch is based on making the customer believe that they are lucky to have been offered the opportunity to beome a Norvergence customer as they are extremely selective. If any technical question where to come up, he was trained to let them know that they where not selected to be part of the program and to move on quickly. I do have a copy of their contract if anyone is interested is taking a look at it. It is an interesting piece of legal work. Michael From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J PozSent: Tuesday, May 04, 2004 9:45 AMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] How does Norvergence do it ? Wow, that was GREAT info... I'm in the NY tri-state area so I'm sure I will run into them as competition. Thanks for sending that link - it's unbelievable; also proves the power of such emailing lists where we can share this type of information about unethical companies!Neil Grant [EMAIL PROTECTED] wrote: Found this:http://w3.ripoffreport.com/reports/ripoff89155.htmMany other nasty stories about them too.--Cheers,Neil-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: 04 May 2004 12:51To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] How does Norvergence do it ?So a guy shows up at the the office, after making anappointment with the office manager / receptionist to talk'phone systems'.After her eyes glaze over, with him talking T1 andFrame-Relay I get to see him. He's from Norvergence. Welldressed. Tells me they can do a T1 for $79, with unlimitedlocal long distance for free. It also does 'internet'.'Just give me copies of your phone bill'. I ask somequestions, like number porting, like provisioning of DIDnumbers, like CIR on the data etc. Now HIS eyes glaze over.That's technical talk ... He's just there to follow up onthe appointment and 'qualify' the customer to see if we areworthy of their cheap service. After I looked at theirwebsite, I can hear 'quack quack'.Cheers,WW Willy WoutersypOne Publishing___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo 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-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!?Win a $20,000 Career Makeover at Yahoo! HotJobs Do you Yahoo!?Win a $20,000 Career Makeover at Yahoo! HotJobs
[Asterisk-Users] would it be possible to...
hi all, i'd like to know if it would be possible with asterisk (and which hardware would i need) to implement the following (or is it not possible with asterisk - but possible with ...) I'd like to set up something like a Mobile to Conventionel Network Gateway - so that users (with there Mobile Phone) which are registered (known Call Number) can Call a Conventionel Network Number + the Number theyed liked to call (for foreign country calls) - the gateway then connects to the foreign number and let the call start. For example: If you'd like to call a number in the united states with your mobile phone (which normally is expensive) - then you call for example 0732/432563-1272626552 (localnumber-number you really like to call) and so you don't have to pay for an expensive foreign call. I hope you understand what i mean (my english isn't best) best regards Wolfgang ___ 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] dialing out to PSTN from SIP phones
On Saturday 01 May 2004 09:42 pm, Tom Scott wrote:
okay, will use ${EXTEN}.
it all seems to be working now. I think my problem was
understanding the flow of control using contexts, but
i also needed to do some reading on syntax and variables
-- and more to come.
the working commands that we ended up using are:
[trunklocal]
exten = _9NX,1,StripMSD(1)
exten = _NX,2,Dial(${TRUNK}/${EXTEN})
exten = _NX,3,Congestion
You could make that example shorter:
exten = _NX,1,Dial(${TRUNK}/${EXTEN:1})
exten = _NX,2,Congestion
Anon
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RE: [Asterisk-Users] How does Norvergence do it ?
I will scan it when I get home tonight and post the url to download it from. Michael From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J Poz Sent: Tuesday, May 04, 2004 10:51 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] How does Norvergence do it ? Yes, I would like to see the contract. Michael Miller [EMAIL PROTECTED] wrote: I have a good friend who used to be a sales rep for them. The entire sales pitch is based on making the customer believe that they are lucky to have been offered the opportunity to beome a Norvergence customer as they are extremely selective. If any technical question where to come up, he was trained to let them know that they where not selected to be part of the program and to move on quickly. I do have a copy of their contract if anyone is interested is taking a look at it. It is an interesting piece of legal work. Michael From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J Poz Sent: Tuesday, May 04, 2004 9:45 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] How does Norvergence do it ? Wow, that was GREAT info... I'm in the NY tri-state area so I'm sure I will run into them as competition. Thanks for sending that link - it's unbelievable; also proves the power of such emailing lists where we can share this type of information about unethical companies! Neil Grant [EMAIL PROTECTED] wrote: Found this: http://w3.ripoffreport.com/reports/ripoff89155.htm Many other nasty stories about them too. -- Cheers, Neil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: 04 May 2004 12:51 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] How does Norvergence do it ? So a guy shows up at the the office, after making an appointment with the office manager / receptionist to talk 'phone systems'. After her eyes glaze over, with him talking T1 and Frame-Relay I get to see him. He's from Norvergence. Well dressed. Tells me they can do a T1 for $79, with unlimited local long distance for free. It also does 'internet'. 'Just give me copies of your phone bill'. I ask some questions, like number porting, like provisioning of DID numbers, like CIR on the data etc. Now HIS eyes glaze over. That's technical talk ... He's just there to follow up on the appointment and 'qualify' the customer to see if we are worthy of their cheap service. After I looked at their website, I can hear 'quack quack'. Cheers, WW Willy Wouters ypOne Publishing ___ 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 Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs
Re: [Asterisk-Users] module help?
On Monday 03 May 2004 01:08 pm, Rich Adamson wrote: I've been running * for eight months in production mode without the init.d/zaptel script in place. Didn't know 'make config' from within the zaptel src directory even existed, and have never seen/heard anyone even mention that before. Its been running fine with a pair of x100p's, however the system is seldom rebooted. Does that imply that * loads the necessary zaptel modules automatically when its started? The modules are loaded at boot time. The automatic script works very nicely. Anon ___ 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] Quality differences of codecs from PRI to SIP
Hello all, I have googled a bit, but was not able to a definite answer (maybe there is not one..) The question is, how different would be the voice qualitiy, if you let translate * from alaw (PRI) to gsm instead of using alaw as codec for sip. And also how would echo and the processor load be affected? The point is, I really would like to use IAX Phone, but is has no alaw codec... (it seems that there is not any win iax client with alaw/mylaw)... I hope you have some ideas and hits Thanks Bye Felix Deierlein ___ 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] would it be possible to...
This is possible with asterisk. There several ways you can do this. You would need a X100P from Digium to interface with the PSTN line coming in. Then you could send the call over VoIP which doesn't require anything more than broadband and a VoIP provider. You should have caller-id on the PSTN line to verify the mobile number. Mobile -- PSTN -- Asterisk -- VoIP -- Foreign Number Best regards, Andrew Kroh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wolfgang Pichler Sent: Tuesday, May 04, 2004 10:53 AM To: Asterisk-Users Mailinglist Subject: [Asterisk-Users] would it be possible to... hi all, i'd like to know if it would be possible with asterisk (and which hardware would i need) to implement the following (or is it not possible with asterisk - but possible with ...) I'd like to set up something like a Mobile to Conventionel Network Gateway - so that users (with there Mobile Phone) which are registered (known Call Number) can Call a Conventionel Network Number + the Number theyed liked to call (for foreign country calls) - the gateway then connects to the foreign number and let the call start. For example: If you'd like to call a number in the united states with your mobile phone (which normally is expensive) - then you call for example 0732/432563-1272626552 (localnumber-number you really like to call) and so you don't have to pay for an expensive foreign call. I hope you understand what i mean (my english isn't best) best regards Wolfgang ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.670 / Virus Database: 432 - Release Date: 4/27/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.670 / Virus Database: 432 - Release Date: 4/27/2004 ___ 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] would it be possible to...
Die GSM Tailnehmer wählen nicht die eigentlich Auslandsnummer - sonder unsere SIP Gateway Nummer + als Durchwahl die Auslandsnummer. Unser SIP Gateway sollte dann die Durchwahl(=Auslandsnummer) wählen und das Gespräch verbinden. So dachte ich mir das auf jeden Fall - obs möglich ist weiß ich nicht genau - deswegen die Frage (es ist mit teurer Switch Hardware auf jeden Fall möglich - eine Firma in Österreich bietet das bereits an) mfG Wolfgang Am Di, den 04.05.2004 schrieb Patrick Stuckenberger um 17:12: wie m?htest du deine GSM Teilnehmer den auf den SIP Gateway bringen? ;-) Mit freundlichen Gr?en / kind regards Patrick S. Stuckenberger Beratung und Entwicklung __ ScaSoft Prozessvisualisierung . EDV-Dienstleistung . it Consulting 6830 Rankweil, Bundesstrasse 102 / Top 4 __ Telefon: +43(0)5522/84245-01, Fax: DW -4 Handy: +43(0)660/84245 01 http://www.scasoft.com/ , [EMAIL PROTECTED] __ Newsflash: 14.12.2003 Er?fnungsfeier der Amberg Ostr?re, Leitsystem und Prozessvisualisierung wurden in der Rekordzeit von 7 Monaten fertigstellt. 11.12.2003 HP Workstation D530, jetzt mit gratis drei Jahre Vort Ort Service und Reaktionszeit innerhalb von 4 Stunden, HP Premium Partner 09.12.2003 Datenleitungsoptimierung zwischen Gendarmerie Bludenz und ABM Hohenems spart dem Land Vorarlberg monatlich EUR 1200,- an Verbindungskosten. anstehende Projekte: 2004 Q1 Skinfit Distributions und Handeslplattform f? 12 L?der 2004 Q1 Gotthardtunnel Leitsystem 2004 Q2 Hotelsystem in KRK 2004 Q2 2way satellite IP Anbindung f? Boden/Tirol [EMAIL PROTECTED] wrote: hi all, i'd like to know if it would be possible with asterisk (and which hardware would i need) to implement the following (or is it not possible with asterisk - but possible with ...) I'd like to set up something like a Mobile to Conventionel Network Gateway - so that users (with there Mobile Phone) which are registered (known Call Number) can Call a Conventionel Network Number + the Number theyed liked to call (for foreign country calls) - the gateway then connects to the foreign number and let the call start. For example: If you'd like to call a number in the united states with your mobile phone (which normally is expensive) - then you call for example 0732/432563-1272626552 (localnumber-number you really like to call) and so you don't have to pay for an expensive foreign call. I hope you understand what i mean (my english isn't best) best regards Wolfgang ___ 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 -- Mit freundlichen Gr?en / kind regards Patrick S. Stuckenberger Beratung und Entwicklung __ ScaSoft Prozessvisualisierung . EDV-Dienstleistung . it Consulting 6830 Rankweil, Bundesstrasse 102 / Top 4 __ Telefon: +43(0)5522/84245-01, Fax: DW -4 Handy: +43(0)660/84245 01 http://www.scasoft.com/ , [EMAIL PROTECTED] __ ___ 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] Pots Extensions
Hi all,
I am either going daft or not reading things right.
I have just installed, (well 10 hours ago) a TDM40B and 3 X100P cards. I
have followed the examples for the conf files to the letter.
I can call the pots extensions OK from IAX clients, SIP clients and from the
incoming X100P cards.
But, if I pick up the handset to make a call all I get is the engaged tone
and the following message.
May 4 23:24:24 WARNING[221200]: pbx.c:1812 ast_pbx_run: Channel 'ZAP/5-1'
sent into invalid extension 's' in context 'default' but no invalid handler.
If I am reading my configs then shouldn't they be going to the internal
context?
Do I need to set-up pots extensions somewhere like IAX Sip extensions?
=
zaptel.conf
fxsks=1-3
fxoks=4-7
loadzone=uk
zapata.conf
signalling=fxs_ks
context=incoming
channel = 1-3
signalling=fxo_ks
context=internal
channel = 4-7
extensions.conf
[internal]
exten = 4090,1,Dial,ZAP/4
exten = 4091,1,Dial,ZAP/5
exten = 4092,1,Dial,ZAP/6
exten = 4093,1,Dial,ZAP/7
exten = _9X.,Dial,ZAP/1,${EXTEN:1}
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RE: [Asterisk-Users] Pots Extensions
Did you put immediate=yes in your zapata.conf? I had similar problems
previously (I have T100p instead of X100p) and it is fixed when I put
immediate=no.
Lisa
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David J
Carter
Sent: Tuesday, May 04, 2004 12:43 PM
To: Asterisk User Group
Subject: [Asterisk-Users] Pots Extensions
Hi all,
I am either going daft or not reading things right.
I have just installed, (well 10 hours ago) a TDM40B and 3 X100P cards. I
have followed the examples for the conf files to the letter.
I can call the pots extensions OK from IAX clients, SIP clients and from
the
incoming X100P cards.
But, if I pick up the handset to make a call all I get is the engaged
tone
and the following message.
May 4 23:24:24 WARNING[221200]: pbx.c:1812 ast_pbx_run: Channel
'ZAP/5-1'
sent into invalid extension 's' in context 'default' but no invalid
handler.
If I am reading my configs then shouldn't they be going to the internal
context?
Do I need to set-up pots extensions somewhere like IAX Sip extensions?
=
zaptel.conf
fxsks=1-3
fxoks=4-7
loadzone=uk
zapata.conf
signalling=fxs_ks
context=incoming
channel = 1-3
signalling=fxo_ks
context=internal
channel = 4-7
extensions.conf
[internal]
exten = 4090,1,Dial,ZAP/4
exten = 4091,1,Dial,ZAP/5
exten = 4092,1,Dial,ZAP/6
exten = 4093,1,Dial,ZAP/7
exten = _9X.,Dial,ZAP/1,${EXTEN:1}
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[Asterisk-Users] Linux IAX client
Folks, It seems like the * v 0.9 and iaxcomm won't speak to each other. Is there another IAX2 client that is usable under Linux (Debian preferred)? Thanks, Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 IAX 17003992910 ___ 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] T1 DID problem
Hello, I have a T1 (not PRI) plugged into my Asterisk server with a T100P card. Everything is working well, except I only get the first digit of the 4 digit DID in Asterisk. The T1 provider (Eschelon) tried switching to 7 digits, and I only got the first digit of the 7. Can anybody help? We're adding another DID and I need to trap it correctly. System info: Asterisk 0.7.2 Zaptel 9.1 Redhat Fedora Core 1 Thanks. Here are snippets from the relevant files: -- zaptel.conf -- span=1,0,0,esf,b8zsem=1-8loadzone=usdefaultzone=us -- extensions.conf -- ; Need an extension to pick up calls from the T1 that uses em wink; This comes in as a 6 instead of 4 full digits; then pass to the s extensionexten = 6,1,Wait(1)exten = 6,2,Goto(incoming,s,1) -- zapata.conf -- [channels] context=incoming signalling=em_w ; rxwink=600 echocancel=yes echotraining=yes group=1 immediate=no channel = 1-8
RE: [Asterisk-Users] Pots Extensions
Lisa
Thanks for that, worked a treat.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Lisa Xie
Sent: 04 May 2004 17:33
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Pots Extensions
Did you put immediate=yes in your zapata.conf? I had similar problems
previously (I have T100p instead of X100p) and it is fixed when I put
immediate=no.
Lisa
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David J
Carter
Sent: Tuesday, May 04, 2004 12:43 PM
To: Asterisk User Group
Subject: [Asterisk-Users] Pots Extensions
Hi all,
I am either going daft or not reading things right.
I have just installed, (well 10 hours ago) a TDM40B and 3 X100P cards. I
have followed the examples for the conf files to the letter.
I can call the pots extensions OK from IAX clients, SIP clients and from
the
incoming X100P cards.
But, if I pick up the handset to make a call all I get is the engaged
tone
and the following message.
May 4 23:24:24 WARNING[221200]: pbx.c:1812 ast_pbx_run: Channel
'ZAP/5-1'
sent into invalid extension 's' in context 'default' but no invalid
handler.
If I am reading my configs then shouldn't they be going to the internal
context?
Do I need to set-up pots extensions somewhere like IAX Sip extensions?
=
zaptel.conf
fxsks=1-3
fxoks=4-7
loadzone=uk
zapata.conf
signalling=fxs_ks
context=incoming
channel = 1-3
signalling=fxo_ks
context=internal
channel = 4-7
extensions.conf
[internal]
exten = 4090,1,Dial,ZAP/4
exten = 4091,1,Dial,ZAP/5
exten = 4092,1,Dial,ZAP/6
exten = 4093,1,Dial,ZAP/7
exten = _9X.,Dial,ZAP/1,${EXTEN:1}
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Re: [Asterisk-Users] T1 DID problem
On Tue, 2004-05-04 at 11:52, Pat Boyle wrote: -- zaptel.conf -- span=1,0,0,esf,b8zs em=1-8 loadzone=us defaultzone=us -- extensions.conf -- ; Need an extension to pick up calls from the T1 that uses em wink ; This comes in as a 6 instead of 4 full digits ; then pass to the s extension exten = 6,1,Wait(1) exten = 6,2,Goto(incoming,s,1) Get that out of your incoming. You have to match on as many of the unique digits they are sending to you. Don't include any other contexts that might match early. Specifically your incoming should probably just contain a list of your DID numbers and a gotos that direct it to the right sections of the dialplan. exten = ,1,goto(Sales-in,s,1) exten = ,1,goto(Tech-in,s,1) exten = ,1,goto(vmail,s,1) exten = ,1,goto(extensions,110,1) exten = ,1,goto(extensions,111,1) Get the picture? With DID you have to be careful not to match too early, and this will help you avoid early matches by only being able to match to the exact DID numbers being sent. -- zapata.conf -- [channels] context=incoming signalling=em_w ; rxwink=600 echocancel=yes echotraining=yes group=1 immediate=no channel = 1-8 -- Steven Critchfield [EMAIL PROTECTED] ___ 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] DSL vs X100P
I was told the X100P will have issues if installed on a line with a DSL connection. Is there a card that will work correctly on a DSL connection? Thanks!!
[Asterisk-Users] Extension Logic Question
I have an extension context that performs an assisted ParkandAnnounce
page. I create a temporary sound file to be played but I would like to
delete it after being used in the page park application. I cant figure
out how to delete the file after it is used in the context
ParkandAnnounce.
Can anyone offer a suggestion?
Thanks,
Kevin
exten = _7,1,Answer
exten = _7,2,Wait(1)
exten = _7,3,Playback(paging)
exten =
_7,4,Playback(/var/spool/asterisk/voicemail/default/${EXTEN:1}/greet
)
exten = _7,5,Playback(presspound)
exten = _7,6,Record(/tmp/pageperson%d:wav)
exten = _7,7,Wait(1)
exten = _7,8,Playback(${RECORDED_FILE}})
exten = _7,9,Wait(1)
exten =
_7,10,ParkAndAnnounce(beep:beep:beep:/var/spool/asterisk/voicemail/d
efault/${EXTEN:1}/greet:${RECORDED_FILE}:hldonext:PARKED|60|Console/dsp|
extensions,${EXTEN:1},1) ^M
exten = _7,11,System(rm ${RECORDED_FILE})
exten = _7,12,Hangup
^
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[Asterisk-Users] multiplle isdn card
Hi to all, I added a second isdn fritz card to my asterisk box to manage a second isdn line. But when I start capi it sees only one controller. How I can enable the second isdn card. Thank you Bye ___ 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] DSL vs X100P
We utilize an X100P on a DSL line provisioned by Verizon with no problems. Just make sure you place the filters in the right place and you wont have any problems. - Brent -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Blackman Sent: Tuesday, May 04, 2004 1:21 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] DSL vs X100P I was told the X100P will have issues if installed on a line with a DSL connection. Is there a card that will work correctly on a DSL connection? Thanks!!
Re: [Asterisk-Users] DSL vs X100P
On Tue, 2004-05-04 at 12:21, John Blackman wrote: I was told the X100P will have issues if installed on a line with a DSL connection. Is there a card that will work correctly on a DSL connection? You were told wrong. There are a FEW people that are having problems with their X100P on a DSL connection. I have at least two X100Ps on two different DSL connections and they work just fine. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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] Can Asterisk support R2 signaling
Hi All: I'm a newbee to Asterisk. I currently working on a project and want to know if Asterisk does support R2 Signaling. Thanks Begra8fl Yes I think so. But you have to download libr2 and compile it, if I am not mistaken. Bart ___ 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] T1 DID problem
Thanks for the reply. If I delete the "6" extension and leave the 6020 extension, asterisk won't answer it and I get the invalid extension message from asterisk. I suspect that for some reason, the zaptel driver is only passing forward "6" of the full four digits "6020." Any thoughts on why I'm only getting a single digit of the 4 digit DID? -Pat On Tue, 2004-05-04 at 11:52, Pat Boyle wrote: -- zaptel.conf -- span=1,0,0,esf,b8zs em=1-8 loadzone=us defaultzone=us -- extensions.conf -- ; Need an extension to pick up calls from the T1 that uses em wink ; This comes in as a 6 instead of 4 full digits ; then pass to the s extension exten = 6,1,Wait(1) exten = 6,2,Goto(incoming,s,1)Get that out of your incoming. You have to match on as many of theunique digits they are sending to you. Don't include any other contextsthat might match early. Specifically your incoming should probably justcontain a list of your DID numbers and a gotos that direct it to theright sections of the dialplan.exten = ,1,goto(Sales-in,s,1)exten = ,1,goto(Tech-in,s,1)exten = ,1,goto(vmail,s,1)exten = ,1,goto(extensions,110,1)exten = ,1,goto(extensions,111,1)Get the picture? With DID you have to be careful not to match too early,and this will help you avoid early matches by only being able to matchto the exact DID numbers being sent. -- zapata.conf -- [channels] context=incoming signalling=em_w ; rxwink=600 echocancel=yes echotraining=yes group=1 immediate=no channel = 1-8-- Steven Critchfield [EMAIL PROTECTED] - Original Message - From: Pat Boyle To: [EMAIL PROTECTED] Sent: Tuesday, May 04, 2004 9:52 AM Subject: T1 DID problem Hello, I have a T1 (not PRI) plugged into my Asterisk server with a T100P card. Everything is working well, except I only get the first digit of the 4 digit DID in Asterisk. The T1 provider (Eschelon) tried switching to 7 digits, and I only got the first digit of the 7. Can anybody help? We're adding another DID and I need to trap it correctly. System info: Asterisk 0.7.2 Zaptel 9.1 Redhat Fedora Core 1 Thanks. Here are snippets from the relevant files: -- zaptel.conf -- span=1,0,0,esf,b8zsem=1-8loadzone=usdefaultzone=us -- extensions.conf -- ; Need an extension to pick up calls from the T1 that uses em wink; This comes in as a 6 instead of 4 full digits; then pass to the s extensionexten = 6,1,Wait(1)exten = 6,2,Goto(incoming,s,1) -- zapata.conf -- [channels] context=incoming signalling=em_w ; rxwink=600 echocancel=yes echotraining=yes group=1 immediate=no channel = 1-8
[Asterisk-Users] Dial zap and music on hold
i tried using music on hold option in the dial command exten = ,1,Dial(zap/1/,20,m) when someone calls me and i picked up the phone, the call will be suddenly dropped. however, if i use a sip client instead of zap (also changing the dial statement to sip), i can answer the incoming call without a problem. is this a known bug? (asterisk cvs 05-03-04 using RedHat v9 on Via mini-ITX) __ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover ___ 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] Error when loading wcfxo
I found similar posts regarding this error but none that answered my question. My zaptel.conf reads: fxsks=1-2 fxoks=3 loadzone=us defaultzone=us and /proc/interrupts: CPU0 0: 5542402 IO-APIC-edge timer 1: 2 IO-APIC-edge keyboard 2: 0 XT-PIC cascade 8: 3 IO-APIC-edge rtc 14: 72582 IO-APIC-edge ide0 16: 81197 IO-APIC-level eth0 17: 54874555 IO-APIC-level wcfxo 20: 54874082 IO-APIC-level wcfxo 22: 54174464 IO-APIC-level wctdm NMI: 0 LOC: 5542842 ERR: 0 MIS: 0 When I run modprobe wcfxo I get the following error: ZT_CHANCONFIG failed on channel 3: No such device or address (6) /lib/modules/2.4.26/misc/wcfxo.o: post-install wcfxo failed /lib/modules/2.4.26/misc/wcfxo.o: insmod wcfxo failed After running modprobe wcfxs, the asterisk machine runs fine and ztcfg doesnt complain. I tried moving cards to different slots but the error didnt go away. If I remark out the fxoks=3 line the error goes away If I reverse the load order in zaptel.conf and fire up the zaptel modules (in reverse), I stop getting the error. So my question is why does the wcfxo try to fire up my TDM400 and is this error a problem?
RE: [Asterisk-Users] DSL vs X100P
If the new FXO doesn't have a filter built in then you will still have to install a filter and might actually still have the same problem. I work with DSL in large quantity on a daily basis... but that makes me far from an expert! :P So I may be wrong... bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of David Creemer Sent: Tuesday, May 04, 2004 1:12 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] DSL vs X100P I seem to be one of the unfortunate ones with this (rare?) problem. Tried two different brand of filters with no luck. I was told by Digium support that the new fxo module for the TDM400P does not have this problem, so I am in the process of switching from an X100P. -- David From: John Blackman [EMAIL PROTECTED] Date: Tue, 4 May 2004 13:21:12 -0400 Subject: [Asterisk-Users] DSL vs X100P I was told the X100P will have issues if installed on a line with a DSL connection. Is there a card that will work correctly on a DSL connection? Thanks!! ___ 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] T1 DID problem
What signaling are you using in /etc/asterisk/zapata.conf (em, em_w, featd)? When I use a DTMF based signaling, I can see the actual DTMF tones as they are received in my 'full' log. Here is an example of what I see (not real phone number) using a signaling type of 'featd': Apr 30 17:02:46 VERBOSE[47121]: -- Starting simple switch on 'Zap/13-1' Apr 30 17:02:46 DEBUG[47121]: DTMF digit: * on Zap/13-1 Apr 30 17:02:46 DEBUG[47121]: DTMF digit: 5 on Zap/13-1 Apr 30 17:02:47 DEBUG[47121]: DTMF digit: 8 on Zap/13-1 Apr 30 17:02:47 DEBUG[47121]: DTMF digit: 2 on Zap/13-1 Apr 30 17:02:47 DEBUG[47121]: DTMF digit: 6 on Zap/13-1 Apr 30 17:02:47 DEBUG[47121]: DTMF digit: 2 on Zap/13-1 Apr 30 17:02:47 DEBUG[47121]: DTMF digit: 3 on Zap/13-1 Apr 30 17:02:47 DEBUG[47121]: DTMF digit: 2 on Zap/13-1 Apr 30 17:02:47 DEBUG[47121]: DTMF digit: 9 on Zap/13-1 Apr 30 17:02:47 DEBUG[47121]: DTMF digit: 5 on Zap/13-1 Apr 30 17:02:48 DEBUG[47121]: DTMF digit: 7 on Zap/13-1 Apr 30 17:02:48 DEBUG[47121]: DTMF digit: * on Zap/13-1 Apr 30 17:02:48 DEBUG[47121]: DTMF digit: 7 on Zap/13-1 Apr 30 17:02:48 DEBUG[47121]: DTMF digit: 0 on Zap/13-1 Apr 30 17:02:48 DEBUG[47121]: DTMF digit: 4 on Zap/13-1 Apr 30 17:02:48 DEBUG[47121]: DTMF digit: 5 on Zap/13-1 Apr 30 17:02:48 DEBUG[47121]: DTMF digit: 9 on Zap/13-1 Apr 30 17:02:48 DEBUG[47121]: DTMF digit: 7 on Zap/13-1 Apr 30 17:02:49 DEBUG[47121]: DTMF digit: 1 on Zap/13-1 Apr 30 17:02:49 DEBUG[47121]: DTMF digit: 2 on Zap/13-1 Apr 30 17:02:49 DEBUG[47121]: DTMF digit: 0 on Zap/13-1 Apr 30 17:02:49 DEBUG[47121]: DTMF digit: 1 on Zap/13-1 So I can see I am being passed *ani*dnis. This might help you track down if you are actually receiving 4 digit dnis. On Tue, 2004-05-04 at 11:05, Pat Boyle wrote: Thanks for the reply. If I delete the 6 extension and leave the 6020 extension, asterisk won't answer it and I get the invalid extension message from asterisk. I suspect that for some reason, the zaptel driver is only passing forward 6 of the full four digits 6020. Any thoughts on why I'm only getting a single digit of the 4 digit DID? -Pat On Tue, 2004-05-04 at 11:52, Pat Boyle wrote: -- zaptel.conf -- span=1,0,0,esf,b8zs em=1-8 loadzone=us defaultzone=us -- extensions.conf -- ; Need an extension to pick up calls from the T1 that uses em wink ; This comes in as a 6 instead of 4 full digits ; then pass to the s extension exten = 6,1,Wait(1) exten = 6,2,Goto(incoming,s,1) Get that out of your incoming. You have to match on as many of the unique digits they are sending to you. Don't include any other contexts that might match early. Specifically your incoming should probably just contain a list of your DID numbers and a gotos that direct it to the right sections of the dialplan. exten = ,1,goto(Sales-in,s,1) exten = ,1,goto(Tech-in,s,1) exten = ,1,goto(vmail,s,1) exten = ,1,goto(extensions,110,1) exten = ,1,goto(extensions,111,1) Get the picture? With DID you have to be careful not to match too early, and this will help you avoid early matches by only being able to match to the exact DID numbers being sent. -- zapata.conf -- [channels] context=incoming signalling=em_w ; rxwink=600 echocancel=yes echotraining=yes group=1 immediate=no channel = 1-8 -- Steven Critchfield [EMAIL PROTECTED] - Original Message - From: Pat Boyle To: [EMAIL PROTECTED] Sent: Tuesday, May 04, 2004 9:52 AM Subject: T1 DID problem Hello, I have a T1 (not PRI) plugged into my Asterisk server with a T100P card. Everything is working well, except I only get the first digit of the 4 digit DID in Asterisk. The T1 provider (Eschelon) tried switching to 7 digits, and I only got the first digit of the 7. Can anybody help? We're adding another DID and I need to trap it correctly. System info: Asterisk 0.7.2 Zaptel 9.1 Redhat Fedora Core 1 Thanks. Here are snippets from the relevant files: -- zaptel.conf -- span=1,0,0,esf,b8zs em=1-8 loadzone=us defaultzone=us -- extensions.conf -- ; Need an extension to pick up calls from the T1 that uses em wink ; This comes in as a 6 instead of 4 full digits ; then pass to the s extension exten = 6,1,Wait(1) exten = 6,2,Goto(incoming,s,1) -- zapata.conf -- [channels] context=incoming signalling=em_w ; rxwink=600 echocancel=yes echotraining=yes group=1 immediate=no channel = 1-8 ___ 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] How does Novergence do it ?
Ahh, just like my momma told me, if it sounds too good to be true, it usually is.. :-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andre Normandin Sent: Monday, May 03, 2004 4:41 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] How does Novergence do it ? I wouldn't believe it until you see it in writing from Norvergence itself! If they indeed can do that for $500/month, pass them over to me, I'd be interested :-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Lance Arbuckle Sent: Monday, May 03, 2004 3:30 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] How does Novergence do it ? I had just about about sold a new asterisk phone system to a local company when they called back asking if I could match a proposal from Novergence.com. I haven't seen anything on paper but was told their proposal was to provide a new phone system that would replace the existing 8 line 12 extension system, provide an internet T-1, unlimited local and long distance, voice mail, and two cellular phones with unlimited nationwide minutes all for the same $500 per month the business is spending now. The internet T-1 would be at least $500 so I'm a bit confused as to how they go about doing this. Does anyone have any details about Novergence and their phone systems and service ??? Thanks, Lance ___ 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] 0.7.2 debs
Tim Sailer wrote: Does anyone still have the 0.7.2 debs hanging around? I need to revert a recent upgrade. We're having too may flaky problems (like softphones being able to dial out fine, but GrandStreams failing to dial every other time), and iaxcomm not working with gsm. Why not diagnose the problem and then assist in solving it, if there really is a problem? Jeremy McNamara ___ 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] 0.7.2 debs
I second this. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jeremy McNamara Sent: Tuesday, May 04, 2004 2:15 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 0.7.2 debs Tim Sailer wrote: Does anyone still have the 0.7.2 debs hanging around? I need to revert a recent upgrade. We're having too may flaky problems (like softphones being able to dial out fine, but GrandStreams failing to dial every other time), and iaxcomm not working with gsm. Why not diagnose the problem and then assist in solving it, if there really is a problem? Jeremy McNamara ___ 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] multiplle isdn card
First thing you must is read next url http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO and if you hav done this, please attach your capi.conf. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de massimo Enviado el: martes, 04 de mayo de 2004 19:31 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] multiplle isdn card Hi to all, I added a second isdn fritz card to my asterisk box to manage a second isdn line. But when I start capi it sees only one controller. How I can enable the second isdn card. Thank you Bye ___ 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] 0.7.2 debs
On Tue, May 04, 2004 at 03:14:43PM -0400, Jeremy McNamara wrote: Tim Sailer wrote: Does anyone still have the 0.7.2 debs hanging around? I need to revert a recent upgrade. We're having too may flaky problems (like softphones being able to dial out fine, but GrandStreams failing to dial every other time), and iaxcomm not working with gsm. Why not diagnose the problem and then assist in solving it, if there really is a problem? Well, there really *is* a problem. I'll try to debug it, but NOT on the client's production system! I need to get them working first. Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 IAX 17003992910 ___ 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] Can Asterisk support R2 signaling
Is it possible to buy some kind of signalling converters from R2 to PRI ? again. please search the archives... this question has been asked answered N*N*N^N times ... no. r2 support in asterisk in far from being complete and it can do only 10% of the work. you can try libr2 from the cvs, but you're on your own. matteo Il mar, 2004-05-04 alle 19:37, Tola Ogunsan ha scritto: Hi All: I'm a newbee to Asterisk. I currently working on a project and want to know if Asterisk does support R2 Signaling. Thanks Begra8fl From: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Asterisk-Users digest, Vol 1 #3647 - 9 msgs Date: Tue, 04 May 2004 13:32:00 -0500 Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re: would it be possible to... (Wolfgang Pichler) 2. Pots Extensions (David J Carter) 3. RE: Pots Extensions (Lisa Xie) 4. Linux IAX client (Tim Sailer) 5. T1 DID problem (Pat Boyle) 6. RE: Pots Extensions (David J Carter) 7. Re: T1 DID problem (Steven Critchfield) 8. DSL vs X100P (John Blackman) 9. Extension Logic Question (Kevin ) --__--__-- Message: 1 Subject: Re: [Asterisk-Users] would it be possible to... From: Wolfgang Pichler [EMAIL PROTECTED] To: Asterisk-Users Mailinglist [EMAIL PROTECTED] Date: Tue, 04 May 2004 18:02:06 +0200 Reply-To: [EMAIL PROTECTED] Die GSM Tailnehmer whlen nicht die eigentlich Auslandsnummer - sonder unsere SIP Gateway Nummer + als Durchwahl die Auslandsnummer. Unser SIP Gateway sollte dann die Durchwahl(=Auslandsnummer) whlen und das Gesprch verbinden. So dachte ich mir das auf jeden Fall - obs mglich ist wei ich nicht genau - deswegen die Frage (es ist mit teurer Switch Hardware auf jeden Fall mglich - eine Firma in sterreich bietet das bereits an) mfG Wolfgang Am Di, den 04.05.2004 schrieb Patrick Stuckenberger um 17:12: wie m?htest du deine GSM Teilnehmer den auf den SIP Gateway bringen? ;-) Mit freundlichen Gr?en / kind regards Patrick S. Stuckenberger Beratung und Entwicklung __ ScaSoft Prozessvisualisierung . EDV-Dienstleistung . it Consulting 6830 Rankweil, Bundesstrasse 102 / Top 4 __ Telefon: +43(0)5522/84245-01, Fax: DW -4 Handy: +43(0)660/84245 01 http://www.scasoft.com/ , [EMAIL PROTECTED] __ Newsflash: 14.12.2003 Er?fnungsfeier der Amberg Ostr?re, Leitsystem und Prozessvisualisierung wurden in der Rekordzeit von 7 Monaten fertigstellt. 11.12.2003 HP Workstation D530, jetzt mit gratis drei Jahre Vort Ort Service und Reaktionszeit innerhalb von 4 Stunden, HP Premium Partner 09.12.2003 Datenleitungsoptimierung zwischen Gendarmerie Bludenz und ABM Hohenems spart dem Land Vorarlberg monatlich EUR 1200,- an Verbindungskosten. anstehende Projekte: 2004 Q1 Skinfit Distributions und Handeslplattform f? 12 L?der 2004 Q1 Gotthardtunnel Leitsystem 2004 Q2 Hotelsystem in KRK 2004 Q2 2way satellite IP Anbindung f? Boden/Tirol [EMAIL PROTECTED] wrote: hi all, i'd like to know if it would be possible with asterisk (and which hardware would i need) to implement the following (or is it not possible with asterisk - but possible with ...) I'd like to set up something like a Mobile to Conventionel Network Gateway - so that users (with there Mobile Phone) which are registered (known Call Number) can Call a Conventionel Network Number + the Number theyed liked to call (for foreign country calls) - the gateway then connects to the foreign number and let the call start. For example: If you'd like to call a number in the united states with your mobile phone (which normally is expensive) - then you call for example 0732/432563-1272626552 (localnumber-number you really like to call) and so you don't have to pay for an expensive foreign call. I hope you understand what i mean (my english isn't best) best regards Wolfgang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE
[Asterisk-Users] Re: How does Novergence do it ?
Tim == Tim Petlock [EMAIL PROTECTED] writes: Tim Be very careful about them. Search the archives of Tim comp.dcom.telecom for details - focus on the last twelve months. Ah, yes. I knew the name sounded familiar. -JimC ___ 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] How to implement configure agents
On Tuesday 04 May 2004 11:37 am, salman khan wrote: Hi I am new to this forum can some body tell me how can i configure and implement agents. if there is any document available on agents implementation plz forward me that http://www.voip-info.org/tiki-index.php?page=Asterisk%20Agents ___ 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] stun server
What is the best free stun server out there? The one that I have looked at from vovida requires two NICs. Is this neccessary? ___ 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] Multiple music's on hold?
On Friday 30 April 2004 10:36 pm, CW_ASN wrote: yes - Original Message - From: Steven Kalcevich [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 01, 2004 7:30 PM Subject: [Asterisk-Users] Multiple music's on hold? Hey there, Is it possible to have multiple music on holds when you run asterisk? Would you (or any other knowledgeable person) be so kind as to give a short, simple example? I searched the Wikki and Google'd the archives without finding an example clearly illustrating how to use multiple, different music on hold's. The closest I can figure is calling SetMusicOnHold before an extension gets dialed, like: exten = 100,1,SetMusicOnHold(SellWidgets) exten = 100,2,Dial(Zap/2,20) and having multiple classes of music on hold defined in musiconhold.conf, like: [classes] default = quietmp3:/var/lib/asterisk/mohmp3 ;loud = mp3:/var/lib/asterisk/mohmp3 ;random = quietmp3:/var/lib/asterisk/mohmp3,-z SellWidgets = quietmp3:/var/lib/asterisk/Sell/ Am I anywhere close to the correct answer? Thanks, Anon ___ 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] stun server
AJ Grinnell wrote: What is the best free stun server out there? The one that I have looked at from vovida requires two NICs. Is this neccessary? Asterisk does not require STUN. Jeremy McNamara ___ 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] stun server
STUN can be nice when connecting to Asterisk behind NAT in some situations. X-Lite/Pro softphones, Grandstream Budgetones and a few other clients make great use of STUN. That being said, the only good (free) STUN server I've seen is the Vovida one that requires two NICs. It works very well, if that is any consolation. Brian --- Jeremy McNamara [EMAIL PROTECTED] wrote: AJ Grinnell wrote: What is the best free stun server out there? The one that I have looked at from vovida requires two NICs. Is this neccessary? Asterisk does not require STUN. Jeremy McNamara ___ 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] g.729 - licenses and opinions
I have a few SIP phones, Cisco 7960s, and was looking into implementing some compression, ala G.729. I'm looking into purchasing a g729 licenses just to get an idea of performance and voice quality, over lans, wireless and single channel isdn. Does anyone have positive/negative experience w/ getting licenses/support from Digium? Hows the sound quality compared w/ g.711? Is 729 better on slow connections? Jitter more/less of a problem then w/ g.711? Was implementation a pain? I've seen the bandwidth comparisons @ http://www.voip-info.org/wiki-Bandwidth+consumption Things look good... if g.729 turns out to be all it perports itself to be then I feel we'd have a real winner. ___ 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] g.729 - licenses and opinions
I have a few SIP phones, Cisco 7960s, and was looking into implementing some compression, ala G.729. I'm looking into purchasing a g729 licenses just to get an idea of performance and voice quality, over lans, wireless and single channel isdn. Does anyone have positive/negative experience w/ getting licenses/support from Digium? Hows the sound quality compared w/ g.711? Is 729 better on slow connections? Jitter more/less of a problem then w/ g.711? Was implementation a pain? I've seen the bandwidth comparisons @ http://www.voip-info.org/wiki-Bandwidth+consumption Things look good... if g.729 turns out to be all it perports itself to be then I feel we'd have a real winner. We've got about five licenses and a remote 7960's v6.3 running over dsl working just fine. The average user cannot tell the difference between 711 and 729. Installation was easy and straight forward, although you'll find comments in the archives that 729 installation requires a non-scsi drive on the * box. In some cases, you might require two licenses even though you might have only a single 729 phone. Think about VM, etc. Error on the side of too many. Can't comment on support; never needed any. ___ 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] A GOOD IP PHONE IAX OR SIP
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi some one can give me information about a good and ship ip phone IAX or SIP Thanks - -- Alvaro Ivan Parres Peredo Director de IT [EMAIL PROTECTED] Tel: (33) 36301294 ~ (33) 36309553 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFAmByaCDkd3lP6lKQRAvWnAJ93b2/Yv6+YmAuGssLz2SuiQdu03QCfQyF0 2satIwN0367cmBzwxjqFOFE= =OrTx -END PGP SIGNATURE- ___ 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] mediatrix 1104
Hi all, I just got a Mediatrix 1104 evaluation unit -- a 4 port fxs sip gateway. There's no printed documentation shipped with the unit, but I have a piece of software for windows that shipped with a different model (which I haven't tried configuring yet), that uses snmp to set misc variables (ip settings, sip stuff, etc.). Fairly baroque interface pretty slim on help... Basically, I'm wondering if anyone's ever configured one of these things for use with *, if anyone could share any tips with me... Doesn't seem like I'm getting it to register w/* -- I thought I'd been setting the proxy username/password in this thing, but I keep getting this with sip debug: to 98.76.54.32:5060 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bKa9fa10127;received=98.76.54.32 From: Port 2 sip:[EMAIL PROTECTED];tag=fd593f07870355f To: Port 2 sip:[EMAIL PROTECTED];tag=as52ef97c9 Call-ID: [EMAIL PROTECTED] CSeq: 1117525281 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=23e26a38 Content-Length: 0 to 98.76.54.32:5060 ast1*CLI Sip read: REGISTER sip:123.45.67.89 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK667022457 Content-Length: 0 To: Port 3 sip:[EMAIL PROTECTED] From: Port 3 sip:[EMAIL PROTECTED];tag=f8e5152d35870bf Call-ID: [EMAIL PROTECTED] CSeq: 1913617706 REGISTER Contact: Port 3 sip:[EMAIL PROTECTED] User-Agent: MxSipApp/4.4.10.60 MxSF/v3.2.6.24 9 headers, 0 lines Using latest request as basis request Sending to 0.0.0.0 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK667022457;received=98.76.54.32 From: Port 3 sip:[EMAIL PROTECTED];tag=f8e5152d35870bf To: Port 3 sip:[EMAIL PROTECTED];tag=as4a4a8cc7 Call-ID: [EMAIL PROTECTED] CSeq: 1913617706 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 98.76.54.32:5060 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK667022457;received=98.76.54.32 From: Port 3 sip:[EMAIL PROTECTED];tag=f8e5152d35870bf To: Port 3 sip:[EMAIL PROTECTED];tag=as4a4a8cc7 Call-ID: [EMAIL PROTECTED] CSeq: 1913617706 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=48e70a35 Content-Length: 0 to 98.76.54.32:5060 ast1*CLI Sip read: REGISTER sip:123.45.67.89 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK4a7fc3bfd Content-Length: 0 To: Port 4 sip:[EMAIL PROTECTED] From: Port 4 sip:[EMAIL PROTECTED];tag=f0384cd93965088 Call-ID: [EMAIL PROTECTED] CSeq: 144760370 REGISTER Contact: Port 4 sip:[EMAIL PROTECTED] User-Agent: MxSipApp/4.4.10.60 MxSF/v3.2.6.24 9 headers, 0 lines Using latest request as basis request Sending to 0.0.0.0 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK4a7fc3bfd;received=98.76.54.32 From: Port 4 sip:[EMAIL PROTECTED];tag=f0384cd93965088 To: Port 4 sip:[EMAIL PROTECTED];tag=as4d58c8ce Call-ID: [EMAIL PROTECTED] CSeq: 144760370 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 98.76.54.32:5060 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK4a7fc3bfd;received=98.76.54.32 From: Port 4 sip:[EMAIL PROTECTED];tag=f0384cd93965088 To: Port 4 sip:[EMAIL PROTECTED];tag=as4d58c8ce Call-ID: [EMAIL PROTECTED] CSeq: 144760370 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=70915041 Content-Length: 0 I'll provide more info, if necessary. Heck, I'll open up my firewall for someone to get into this mediatrix fiddle with it if they want... Thanks, Jeremy Jones ___ 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] stun server
I just put multiple IPs on the same interface and use -a eth0:1 ip. Seems to work 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
Re: [Asterisk-Users] mediatrix 1104
I just got a Mediatrix 1104 evaluation unit -- a 4 port fxs sip gateway. There's no printed documentation shipped with the unit, but I have a piece of software for windows that shipped with a different model (which I haven't tried configuring yet), that uses snmp to set misc variables (ip settings, sip stuff, etc.). Fairly baroque interface pretty slim on help... Basically, I'm wondering if anyone's ever configured one of these things for use with *, if anyone could share any tips with me... Doesn't seem like I'm getting it to register w/* -- I thought I'd been setting the proxy username/password in this thing, but I keep getting this with sip debug: Seems all of the Mediatrix stuff is configured through snmp only. Finding and changing the parameters is a royal pain, however others have posted to the list using that same model. I would stay away from their fxo model however. After many hours of working with a reseller, ended up having to send it back. Mediatrix's gameplan seems to be oriented towards selling the fxs and fxo boxes in pairs as a form of toll bypass. They really aren't interested in standards and making their products work with *, etc. ___ 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 12SP+
Hi Paul, To my knowledge, you can't change the image on them. I recently bought 3 of them, and we help from this list, I was able to connect them to my asterisk server. However, they are not fully functional. I can make calls and hear calls, but I'm muted. I'm looking for a solution. The protocol they use is skinny, which I don't think is complete. My suggestion is to avoid them for now. -Ry On Sun, 2004T-05-02 at 16:00, Paul Tyreman wrote: Hi, I'm thinking about getting a couple of Cisco 12SP+ phones to use on my Asterisk system. I have just bought a Cisco 7960, and they are great, but too expensive to buy a lot of them, so I though I might try the 12SP+ ones. I have seen in the archives that the phones work on Asterisk, but I can't see much in there about the images in use. When I got my 7960, it had the call manager image on it, and I had to convert it to the SIP image before I could use it. Is this the same case with the 12SP+, do you need to change it's image ? Thanks in advance, Paul. ___ 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] Dial zap and music on hold
On Tuesday 04 May 2004 06:13 pm, Jet Bagadion wrote: i tried using music on hold option in the dial command exten = ,1,Dial(zap/1/,20,m) Did you mean exten = ,1,Dial(zap/1,20,m) ? Anon ___ 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] mediatrix 1104
Rich et alia, Seems all of the Mediatrix stuff is configured through snmp only. Finding and changing the parameters is a royal pain, Yer tellin' me! however others have posted to the list using that same model. Really? I wasn't able to come up with anything googling, other than someone else asking how to configure the things... Please, throw up a link if you see something I don't. I would stay away from their fxo model however. After many hours of working with a reseller, ended up having to send it back. I'm on the verge with this one. Mediatrix's gameplan seems to be oriented towards selling the fxs and fxo boxes in pairs as a form of toll bypass. They really aren't interested in standards and making their products work with *, etc. But one would think it'd be fairly simple to at least do a straightforward sip proxy registration, no? Anyhoo -- I'll beat on it now then for a couple days post results if anyone's interested. In the meantime, my offer to open up access to anyone who'd like to take a stab at it is still on the table. Thanks, Jeremy ___ 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 12SP+
Ryan Laginski ([EMAIL PROTECTED]) wrote: Hi Paul, To my knowledge, you can't change the image on them. I recently bought 3 of them, and we help from this list, I was able to connect them to my asterisk server. However, they are not fully functional. I can make calls and hear calls, but I'm muted. I'm looking for a solution. The protocol they use is skinny, which I don't think is complete. My suggestion is to avoid them for now. -Ry Hi ry, hi all, just to give you an overview of what the problem with the 12SP+ is and our plan to support these in chan_sccp(experimental) (*1): - the 12SP+/30VIP are non intelligent phones compared to the 7960 and/or 7920 phones. - every keypress is being transferred through the skinny protocol to the respective server. - i have a 12SP+ to test things with (thanks for the contributor) I think i will have a somehow better working driver support in about 1 month from now on. we'll see... --jan *1 http://chan-sccp.sf.net ___ 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] Can Asterisk support R2 signaling
Bartosz Jozwiak wrote: Hi All: I'm a newbee to Asterisk. I currently working on a project and want to know if Asterisk does support R2 Signaling. Thanks Begra8fl Yes I think so. But you have to download libr2 and compile it, if I am not mistaken. You are mistaken. Regards, Steve ___ 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] New ENUM service, what do you think?
At 7:14 PM +1000 on 5/4/04, Duane wrote: John Todd wrote: TRIP (RFC 3219) is the answer, but I'm the only one pounding that drum, it seems. If anyone here on the list has $100,000 to put together a real programming effort towards getting that implemented, y'all let me know. The longer this waits, the more lame and broken become the solutions offered. sigh One small oversight in your thinking, something like TRIP will only benefit large telcos and VOIP providers with interconnects, I don't see this flowing down to a tangible benefit to the average person, where as something like enum.164 is. TRIP is based on BGP and BGP already does most of the IP routing smarts TRIP is supposed to be beneficial for, however that $100k would be better spent on improving the smarts in the call routing software rather then turning things back into a hub and spoke model, p2p is way more efficient if it can be utilised to it's full potential. At this stage the only potential method to prevent VOIP spam is something like SPF records, which would only end up duplicate enum. It's a lot harder to get phone numbers then IP addresses, so this would overcome people's concerns about dynamically allocated IPs, phone numbers aren't. -- Best regards, Duane I strongly disagree with your summary that TRIP doesn't help the smaller user. In fact, the reason I'm so strongly an advocate of some type of TRIP development is that it removes the barriers for small entities in the pursuit of better call rates for TDM offload and VoIP interconnection. Comparative routing data should not be the sole domain of huge telephony firms. One example... Currently, I see quite a few people here trying to get good rates to various international destinations (regardless of their nation of origin.) Wouldn't it be nice to have a protocol that allowed the home or small business user to have COMPETING long distance carriers on a per-call basis? When one of them runs a sale, your voice traffic could (according to your rules) shift over to the least expensive/best sounding/whatever carrier that you'd chosen. Just get a TRIP feed from three or four carriers, and away you go. It all would happen automatically, and you could preference or de-preference certain metrics as you went along but the carriers will be sending you their most up-to-date routing information for PSTN handoff destinations. Wouldn't it be great if your Asterisk server had that ability? This is just one use and benefit case of TRIP; there are many others. If you say that ENUM is going to solve that problem by offering pointers for every phone prefix in the world in the next 5 years, or even 33% of them, I would suggest that is a rather optimistic outlook. ENUM cannot have competing answers to the same question; it MUST have a single answer, no matter how many private ENUM servers you put in the path (otherwise, you're just redesigning TRIP.) TDM offload in between VoIP networks is here to stay; we just need a protocol that allows inter-system route exchange for those of us lucky enough to be able to take advantage of it today, not sometime in the far off future. Yes, it will also help large carriers as well for their exchange of route information, but it's not limited to their use. TRIP is like BGP in it's design, but extremely different in it's implementation. It layers on top of IP, so arguments comparing BGP to TRIP with terms like hub and spoke are invalid. Destination information does not (necessarily) follow any of the path of the lower layers of the routing protocol. Additionally, I am unclear on how you believe that TRIP is involved in IP routing smarts. The two are not linked in any way. Can you clarify? I am uncertain to what your final comments about spam refer. Neither ENUM nor TRIP address issues of call validation in a realistic manner; any SPF-like methods for verifying origination work equally well with either reference scheme. Remember that ENUM is a stopgap, and we should do all we can to move away from numbers as an addressing scheme for VoIP (or any protocol) delivery. My SIP phone address is [EMAIL PROTECTED] but the only reason most people can't use that is because they are crippled by phones with numeric keypads. ENUM is the in-between method to map numbers to more flexible addressing until we have smarter phones on our desks and we can use the more flexible addressing methods to dial the other party. As I've said, I am a firm believer in ENUM as a second-generation VoIP routing method, but I'm just as firm a believer (due to very hard-won experience in the PBX and carrier markets) that it is insufficient at this time to make any difference at all in anything other than the most theoretical environments, or environments that have been jury-rigged to use ENUM because there was nothing better available. JT ___ Asterisk-Users mailing
RE: [Asterisk-Users] grandstream transfer, park and conference
1. Check if Asterisk is always in the media path, i.e. you need the t or T option (or something similar) in your Dial statement. Alternatively you could introduce a canreinvite=no in sip.conf for the GS phones. 2. Check your context setup in extensions.conf and make sure that in call cases your GS phone has the parkedcalls context available Philipp I have an update for this problem, and I discovered strange problems. I can do transfer, call parking nicely now except one thing: * only recognize one dtmf only (for example when I press # on my budgetone, it will say transferring, and put my caller on music, but when I press 234, * only catch 2 (in my budgetone, it will say there's no valid extension .), but if I transfer it to one digit extension first (when the call is received, then I want to do transferring/parking/meetme, I need to transfer the call to extension that has only 1 digit, then it will work perfectly (I can transfer anywhere I want (2/3/4 digits)) Is it a bug? If it is, from budgetone, or *? And how to deal with it? Thanks Isianto ___ 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] DSL vs X100P
I am trying to forward an inbound call to go out through another X101P and I get nothing but a noise like a helicopter sound... Inbound and outbound are ok if done separately. I already checked IRQs and they are fine. Updated the drivers and asterisk and they seem to be ok too. Turned on and off echo cancel. Both lines are coming from an ISDN line,channels A and B respectively. Should it be cable problem or another issue, in this case with ISDN lines? Isamar ___ 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] New ENUM service, what do you think?
John Todd wrote: I strongly disagree with your summary that TRIP doesn't help the smaller user. In fact, the reason I'm so strongly an advocate of some type of TRIP development is that it removes the barriers for small entities in the pursuit of better call rates for TDM offload and VoIP interconnection. Comparative routing data should not be the sole domain of huge telephony firms. Call rates not calls... Sure the PSTN network is still the most widely used for voice calls now, but for how much longer? I'm currently in the process of doing a feasibility study and roll out to a large number of offices in Australia to route calls between offices using their existing DSL connections. Currently the main carrier is going round trying to sign everyone up on exclusivity contracts for 2 years for all voice calls, for 16c per minute... TRIP won't help there... have competing answers to the same question; it MUST have a single answer, no matter how many private ENUM servers you put in the path Erm no, we're already working out patches for asterisk to deal with multiple answers, including dealing with tel fields in a sane manner... The whole purpose of TRIP is to route calls via the equivalent of carriers, how many of those carriers will let you add your voip records to their database and take revenue away from them? TRIP is all about centralised control away from the end user, while it might give them short term benefits in being able to save a few dollars here and there long term they will be locked into using carriers for internet to internet calls that they could be making for free. While enum doesn't have the ability to make cost decisions directly what you are suggesting would require everyone to sign up with all providers, or have a shared database of user details or some where in between and wouldn't that leave the end user open to being slammed? spammed? or many other things by companies trying to get ahead? Like many things in this world they all would work perfectly in theory, in practise they end up being abused till people get sick of it and just walk away to a simpler system. layers of the routing protocol. Additionally, I am unclear on how you believe that TRIP is involved in IP routing smarts. The two are not linked in any way. Can you clarify? Sure, internet to internet calls are already paid for in the leasing of bandwidth, why pay a phone company to route the call via IP for you when it could be done at no additional cost? I am uncertain to what your final comments about spam refer. Neither ENUM nor TRIP address issues of call validation in a realistic manner; any SPF-like methods for verifying origination work equally well with either reference scheme. Remember that ENUM is a stopgap, and we should do all we can to move away from numbers as an addressing scheme for VoIP I don't think it was designed as a stop gap, more likely as a method of more easily tracking people with public records that didn't need a search warrant to access them... TRIP is I see it, is a method of routing calls more then working out where to send the call to directly. enum points out specifically where the call should go and could be used in reverse to find out where the call should have come from. (or any protocol) delivery. My SIP phone address is [EMAIL PROTECTED] but the only reason most people can't use that is because they are crippled by phones with numeric keypads. ENUM is the in-between method to map numbers to more flexible addressing until we have smarter phones on our desks and we can use the more flexible addressing methods to dial the other party. I don't want something as large as a small laptop to lug around to make phone calls with, if you ever do a reasonable amount of SMS'ing you will learn how much of a pain in the a** that can be. Using a single number as a point of reference to all the contact information on a company or a person within a company would be very useful to me. To send an email I could use his enum number, to contact him via icq I could use his enum number, to make a phone call I could use his enum number, to fax him I could use his enum number then have the fax machine lookup his email address and route the image via that instead. As I've said, I am a firm believer in ENUM as a second-generation VoIP routing method, but I'm just as firm a believer (due to very hard-won experience in the PBX and carrier markets) that it is insufficient at this time to make any difference at all in anything other than the most theoretical environments, or environments that have been jury-rigged to use ENUM because there was nothing better available. From your email you are hinting that TRIP is a stop gap measure between pure internet telephony and the PSTN network, I'm suggesting enum is a longer term point to point method, while it may seem stop gap in a hybrid system long term it will be the best method of the 2, if you don't
Re: [Asterisk-Users] DSL vs X100P
Isamar Maia wrote: I am trying to forward an inbound call to go out through another X101P and I get nothing but a noise like a helicopter sound... Inbound and outbound are ok if done separately. I already checked IRQs and they are fine. Updated the drivers and asterisk and they seem to be ok too. Turned on and off echo cancel. Both lines are coming from an ISDN line,channels A and B respectively. Should it be cable problem or another issue, in this case with ISDN lines? I had a similar problem after a CVS update and had to set the rxgain to -2 to reduce the time the echo canceller kicked in... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers ___ 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] Extension Logic Question Help!! Park and Announce
I have an extension context that performs an assisted ParkandAnnounce
page. I create a temporary sound file to be played but I would like to
delete it after being used in the page park application. I cant figure
out how to delete the file after it is used in the context
ParkandAnnounce.
Can anyone offer a suggestion?
Thanks,
Kevin
exten = _7,1,Answer
exten = _7,2,Wait(1)
exten = _7,3,Playback(paging)
exten =
_7,4,Playback(/var/spool/asterisk/voicemail/default/${EXTEN:1}/greet
)
exten = _7,5,Playback(presspound)
exten = _7,6,Record(/tmp/pageperson%d:wav)
exten = _7,7,Wait(1)
exten = _7,8,Playback(${RECORDED_FILE}})
exten = _7,9,Wait(1)
exten =
_7,10,ParkAndAnnounce(beep:beep:beep:/var/spool/asterisk/voicemail/d
efault/${EXTEN:1}/greet:${RECORDED_FILE}:hldonext:PARKED|60|Console/dsp|
extensions,${EXTEN:1},1) ^M
exten = _7,11,System(rm ${RECORDED_FILE})
exten = _7,12,Hangup
^
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Re: [Asterisk-Users] Linux IAX client
On Tue, 4 May 2004 12:32:30 -0400, Tim Sailer [EMAIL PROTECTED] wrote: Folks, It seems like the * v 0.9 and iaxcomm won't speak to each other. Is there another IAX2 client that is usable under Linux (Debian preferred)? Thanks, Tim Did it work before you upgraded asterisk, or you can't get it to work at all? I'll admit that the QUICKSTART is a bit terse. ___ 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] Dial zap and music on hold
didn't encounter the sudden call hangup when i add Answer before that. exten = ,1,Answer exten = ,2,Dial(zap/1,20,m) On Tuesday 04 May 2004 06:13 pm, Jet Bagadion wrote: i tried using music on hold option in the dial command exten = ,1,Dial(zap/1/,20,m) Did you mean exten = ,1,Dial(zap/1,20,m) ? Anon __ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover ___ 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] vonage sip url
Hello List, anybody knows the sip url of vonage ??? like [EMAIL PROTECTED] ?? regards. -Neo ___ 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] DSL vs X100P
I had a similar problem after a CVS update and had to set the rxgain to -2 to reduce the time the echo canceller kicked in... The problem is that my settings now only work well with rxgain=+15 txgain=+15 Setting rxgain to -10, the noise disappeared but I can hear only one side of the line. Isamar [EMAIL PROTECTED] Nagoya/Japan ___ 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
