Re: [Asterisk-Users] static database config gui
Before I start on this next step I want to see if my plan has any flaws. I'm working on adding multi user functionality to the gui. By multi user I mean the ability of any number of end users to create their own configurations on the same server, without interferring with other users. The only thing I can think of that will work is to prefix context names, section names, and globals with an id unique to the user (most likely their username). Then I could do the translation transparent to the user so they only see the name without the prefix. The trick as I see it is to come up with the best separator for the prefixed context/section names. It needs to be something that won't collide with anything else, and that asterisk will accept as a name. Anyone have an opinion on using a double underscore for this? For example context [mycontext] for user chris would become [__chris__mycontext__]. To avoid collisions I could then make the double underscore an illegal character combination anywhere else. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] capi dial in/out configuration
Hi
all,
I've
recentrly starting to play around with *, when all I wanted is to configure an
fritz ISDN card with [EMAIL PROTECTED]
Currently
I'm stuck at the phase of what do I do with capi after everything is installed.
I'm
trying to understand how to setup incoming and outgoing calls at [EMAIL PROTECTED] since I'm
getting a bit lost with the default dial plan.
It
seems that * answers but disconnect it directly, and I'm unable to setup
outgoing calls.
I
know this is a very general question, but if anyone could give me some pointers
about how to setup capi dial plan, and explain some terms like msn in the
capi.conf file.
My
capi.conf
[EMAIL PROTECTED] asterisk]# cat
capi.conf |grep -v ';'
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=50
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=demo
devices=2
I've
added these two lines the extensions_custom:
s,1,Dial,CAPI/@50:b${EXTEN}|30
always early B3
s,1,Dial,CAPI/@50:${EXTEN}|30|r
no early B3, fake ring indication
when
dialing out I get:
--
Executing Macro("SIP/200-3b6b", "dialout-trunk|1|999") in
new stack
-- Executing
GotoIf("SIP/200-3b6b", "fooOhad?4") in new stack
-- Executing
SetCallerID("SIP/200-3b6b", "Ohad Levy") in new stack
-- Executing
Goto("SIP/200-3b6b", "6") in new stack
-- Goto (macro-dialout-trunk,s,6)
-- Executing
SetGroup("SIP/200-3b6b", "OUT_1") in new stack
-- Executing
CheckGroup("SIP/200-3b6b", "") in new stack
-- Executing
SetVar("SIP/200-3b6b", "DIAL_NUMBER=999") in new stack
-- Executing
SetVar("SIP/200-3b6b", "DIAL_TRUNK=1") in new stack
-- Executing
AGI("SIP/200-3b6b", "fixlocalprefix") in new stack
-- Launched
AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script
fixlocalprefix completed, returning 0
-- Executing
Dial("SIP/200-3b6b", "/999") in new stack
== Everyone is busy/congested
at this time
-- Executing
NoOp("SIP/200-3b6b", "dial failed") in new stack
-- Executing
Macro("SIP/200-3b6b", "outisbusy") in new stack
-- Executing
Playback("SIP/200-3b6b", "allison7/all-circuits-busy-now") in
new stack
-- Playing 'allison7/all-circuits-busy-now'
(language 'en')
== Spawn extension (macro-outisbusy,
s, 1) exited non-zero on 'SIP/200-3b6b' in macro 'outisbusy'
== Spawn extension (from-internal,
, 2) exited non-zero on 'SIP/200-3b6b'
-- Executing
Macro("SIP/200-3b6b", "hangupcall") in new stack
-- Executing
ResetCDR("SIP/200-3b6b", "w") in new stack == Starting CAPI[contr1/8856224]/0
at demo,8856224,1 failed so falling back to exten 's'
== Starting CAPI[contr1/8856224]/0
at demo,s,1 still failed so falling back to context 'default'
-- Executing
Playback("CAPI[contr1/8856224]/0", "vm-goodbye") in new
stack
-- started
pbx on channel (callgroup=0)!
-- Playing 'vm-goodbye'
(language 'en')
-- Executing
Macro("CAPI[contr1/8856224]/0", "hangupcall") in new stack
-- Executing
ResetCDR("CAPI[contr1/8856224]/0", "w") in new stack
-- Executing
NoCDR("CAPI[contr1/8856224]/0", "") in new stack
-- Executing
Wait("CAPI[contr1/8856224]/0", "5") in new stack
-- Executing
Hangup("CAPI[contr1/8856224]/0", "") in new stack
== Spawn extension (macro-hangupcall,
s, 4) exited non-zero on 'CAPI[contr1/8856224]/0' in macro 'hangupcall'
== Spawn extension (default,
s, 2) exited non-zero on 'CAPI[contr1/8856224]/0'
-- Executing
NoCDR("SIP/200-3b6b", "") in new stack
-- Executing
Wait("SIP/200-3b6b", "5") in new stack
== Spawn extension (macro-hangupcall,
s, 3) exited non-zero on 'SIP/200-3b6b' in macro 'hangupcall'
== Spawn extension (from-internal,
h, 1) exited non-zero on 'SIP/200-3b6b'
When
receiving a call:
==
Starting CAPI[contr1/]/0 at demo, ,1 failed so
falling back to exten 's'
== Starting CAPI[contr1/]/0
at demo,s,1 still failed so falling back to context 'default'
-- Executing
Playback("CAPI[contr1/]/0", "vm-goodbye") in
new stack
-- started
pbx on channel (callgroup=0)!
-- Playing 'vm-goodbye'
(language 'en')
-- Executing
Macro("CAPI[contr1/]/0", "hangupcall") in
new stack
-- Executing
ResetCDR("CAPI[contr1/]/0", "w") in new
stack
-- Executing
NoCDR("CAPI[contr1/]/0", "") in new stack
-- Executing
Wait("CAPI[contr1/]/0", "5") in new stack
-- Executing
Hangup("CAPI[contr1/]/0", "") in new stack
== Spawn extension (macro-hangupcall,
s, 4) exited non-zero on 'CAPI[contr1/]/0' in macro 'hangupcall'
== Spawn extension (default,
s, 2) exited non-zero on 'CAPI[contr1/]/0'
Thanks
a lot,
Ohad
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Re: [Asterisk-Users] static database config gui
On 5/26/05, Magnus Espeland <[EMAIL PROTECTED]> wrote: > Hi, > > Nice work! > > Is it easy to make it run without mod_perl? > No it would take a lot of changes to make it work without mod perl. I thought a lot about converting it to a regular perl cgi. The thing is it's a trade off between being a bit easier to install if mod perl is required versus quite a significant performance hit without mod perl. What I will do is make the install as easy as possible. I'm thinking about bundling all the required modules including the Template Toolkit, so that except for mod perl you shouldn't have to install anything not included with the distribution. > Is the source online? Not yet. I need to take what I have, clean it up a bit, and write up an installation guide. Hopefully in a week I should have it ready. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] International Caller ID?
Anytime I receive a landline to anything over here in AUS, it comes up as Overseas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Malcolm-Smith Sent: Friday, 27 May 2005 3:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] International Caller ID? Rod Bacon wrote: > We have antiquated caller ID schemes here in Australia. We barely > support numbers from other local carriers, let alone OS ones. > Certainly no names either. When dialing out thru voipjet, I can put anything I like and it will come thru to my mobiles in New Zealand just fine (on both networks) - However calls to landlines just come up as on the caller ID as they put that for any international call. -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.17 - Release Date: 25/05/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.17 - Release Date: 25/05/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P vs SIP3000 x2
My vote is for the Sipura as well; essentially I agree with all of Brian's points. Those boxes work well once set up correctly. And, the server and PSTN line doesn't even have to be close; PSTN line in Washington, * server in Texas and a slow (as in 128 kbps) DSL between them is sufficient. There are two "issues" that I can think of: 1) as an outgoing FXO interface you do not get any call progress; the Sipura will answer the call right away and pass down the PSTN audio; it does not do busy detect or the like on outgoing calls. For incoming calls you can have * when to answer the call, if at all. Call disconnection detection and the like works quite well once tuned (which isn't always trivial). 2) Indeed there is low audio volume on *SOME* calls. In my experience it always occurs with the same callers, so it's somehow related with the other end. Personally I did not find it to be an issue (the conversation can be understood just fine) except voicemail tends to detect silence when there is none in these cases. The recorded part is actually fine, low volume but quite understandable. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] International Caller ID?
Rod Bacon wrote: We have antiquated caller ID schemes here in Australia. We barely support numbers from other local carriers, let alone OS ones. Certainly no names either. When dialing out thru voipjet, I can put anything I like and it will come thru to my mobiles in New Zealand just fine (on both networks) - However calls to landlines just come up as on the caller ID as they put that for any international call. smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do I diagnose the problem in this Asterisk test session with FWD?
> Call rejected: 503 Service Unavailable The above is quite correct. You will notice that there is no answer from FWD whatsoever to the NEW or REGREQ packets. Either your firewall is blocking IAX traffic (incoming or outgoing), FWD's IAX is down or unreachable from your location due to network trouble. "iax2 show registry" probably shows "Request Sent". My suggestion, try again later or use tcpdump to check for IAX traffic on your interface (like "tcpdump -n src or dst port 4569"). --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] does Jitter calculation in chan_iax2.c work???
Hi, We are trying to get the jitter of a channel for iax channels. iax2 show netstats The above command always shows zero value for jitter. (Actually, only rtt and kpkts are non-zero). The behaviour is the same even for cross-continental calls. Is this a bug in the implementation or a configuration problem?. Thanks, Vijay & Ashish PS:We have enabled jitterbuffer at both ends. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk crashes with sipp
with asterisk running, if i call sipp -s 9111 -d 6 -r 20 -t un -sn uac -m 60 all the calls get set up, and after a minute when asterisk receives the 1st BYE from uac, it responds with 200 OK and then crashes. If i restart asterisk, all the calls get terminated properly. in the extensions, i have [default] exten => 9111222,1,Answer exten => 9111222,3,Wait(600) exten => 9111222,4,Hangup please help as i am unable to continue with any load tests ! tulika _ Kareena or Rani? Saif or SRK? http://server1.msn.co.in/sp05/iifa/ Rock your vote now at IIFA. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sipsak with asterisk
i am attaching the trace of the sipsak error when run with the command below. # sipsak -UI -a password -s sip:[EMAIL PROTECTED]:5060 - warning: ignoring -i option when in usrloc mode fqdnhostname: 127.0.0.1 username: 985389744 domainname: 203.197.212.211:5060 request: REGISTER sip:203.197.212.211:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:1026;rport From: sip:[EMAIL PROTECTED]:5060;tag=3b60505e To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:1026 Expires: 15 Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.8.12 registering user 985389744... ignoring provisinal response authorizing received: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 127.0.0.1:1026;received=203.197.212.211;rport=1026 From: sip:[EMAIL PROTECTED]:5060;tag=3b60505e To: sip:[EMAIL PROTECTED]:5060;tag=as72f0756d Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: WWW-Authenticate: Digest realm="asterisk", nonce="5556cf14" Content-Length: 0 registering user 985389744... ignoring provisinal response OK SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:1026;received=203.197.212.211;rport=1026 From: sip:[EMAIL PROTECTED]:5060;tag=3b60505e To: sip:[EMAIL PROTECTED]:5060;tag=as72f0756d Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 15 Contact: ;expires=15 Date: Thu, 26 May 2005 04:03:58 GMT Content-Length: 0 username: 985389744 domainname: 203.197.212.211:5060 ack: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:1026;rport From: sip:[EMAIL PROTECTED]:1026;tag=4c37a25f To: sip:[EMAIL PROTECTED]:5060;tag=11415721137 Call-ID: [EMAIL PROTECTED] CSeq: 2 ACK Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.8.12 reply: SIP/2.0 200 OK From: sip:[EMAIL PROTECTED]:1026;tag=4c37a25f To: sip:[EMAIL PROTECTED]:5060;tag=11415721137 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Content-Length: 0 User-Agent: sipsak 0.8.12 request: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:1026;rport From: sip:[EMAIL PROTECTED]:1026;tag=4c37a25f To: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Content-Length: 0 Contact: sip:[EMAIL PROTECTED]:1026 Subject: DONT ANSWER this test call! Max-Forwards: 70 User-Agent: sipsak 0.8.12 inviting user 985389744... authorizing received: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 127.0.0.1:1026;received=203.197.212.211;rport=1026 From: sip:[EMAIL PROTECTED]:1026;tag=4c37a25f To: sip:[EMAIL PROTECTED]:5060;tag=as680d9091 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="0469ee64" Content-Length: 0 error: could not find To in the reply -- tulika From: "Tulika Pradhan" <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] sipsak with asterisk Date: Fri, 13 May 2005 05:55:01 + i am using sipsak to test asterisk. i use the command $ sipsak -U -I -s sip:[EMAIL PROTECTED] -vv -x 1800 -e3 and i get the message SIP/2.0 407 Proxy Authentication Required as a response to INVITE message (REGISTER was successful) and error: could not find To in the reply does anyone have some idea as to what is missing ? tulika _ Kareena or Rani? Saif or SRK? http://server1.msn.co.in/sp05/iifa/ Rock your vote now at IIFA. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ The MSN Gamezone! http://www.msn.co.in/gamezone Ready for the challenge? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Using zap channels on 2 different servers
Be sure to have the proper digit matching order in extensions.conf.
I currently have local lines (all US National) going through one * with a
single T1 and long distance going through a * server with a quad. Based upon
the number dialed, assuming 9 is the prefix, the number gets routed
appropriately from a third * server that SIP clients log into.
So from the * with the SIP clients
Exten => _9XX,1,blah
And
Exten => _9XX,1,blah
Are in a specific order so the first match "wins"
I hope this helps.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson
Sent: Thursday, May 26, 2005 11:25 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Using zap channels on 2 different servers
You use pattern matching in extensions.conf to determine where the call is
to go. In North America, you dial 1+3+3+4 digits for LD and 3+4 for local.
In Europe, you dial 3+3+3 between countries or 3+3 in the same country* ,
so,
EUROPEAN SERVER:
[outbound]
exten => _9XXX,1,Dial(IAX2/europeuser:[EMAIL PROTECTED]/${exten})
'Local north america calling 3+4 digit
exten =>
_9XXX,1,Dial(IAX2/europeuser:[EMAIL PROTECTED]/${exten}) 'LD
north america calling 1+3+3+4 digit)
AMERICA SERVER:
[outbound]
exten => _9XX,1,Dial(IAX2/americauser:[EMAIL PROTECTED]/${exten})
'Local europe calling 3+3 digit
exten =>
_9X,1,Dial(IAX2/americauser:[EMAIL PROTECTED]/${exten})
'Cross-country europe calling 3+3+3 digit)
In this example, the X's mean any digit. If you count the number of X's, you
get 7 digits for NA local, 11 for LD, 6 for European local, 9 for European
LD.
In both of these examples, dialing 9 before the number is assumed. Also, it
is assumed that IAX.CONF on both servers is set up correctly to dump the
inbound IAX call into a context that has access to the ZAP channels of the
respective server.
You could also modify this so if they dial 8 first, for example, it would
always go to Europe or vice versa. But then it becomes a user training
issue. If you just tell them to dial 9 + the number then Asterisk will
figure out the rest.
hth
*It's my understanding that there are exceptions in certain European
countries. You will have to figure out those execptions.
-Original Message-
From: Martin Roy [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 26, 2005 11:49 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Using zap channels on 2 different servers
Let say I have a server located in Europe and one in North America.
The 2 servers are connected together with iax2.
Both server are connected to phone lines in there own country.
If I want that when a user call a north american phone number from the
server in Europe it use a zap channel on the server located in North America
and also if someone in North America dial an European phone number it use a
zap channel on the server in Europe how can I setup this?
Currently it's configure that when a user press 9 and enter the number to
dial it takes a local zap channels to dial out.
I figured easily how to call a remote extension as I use a different range
of extensions for both site but can I use remote zap channels the same way?
Thanks
Martin
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--
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Checked by AVG Anti-Virus.
Version: 7.0.322 / Virus Database: 266.11.17 - Release Date: 5/25/2005
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RE: [Asterisk-Users] International Caller ID?
Hi Nathan, I'm an aussie in New York, on my incoming Packet 8 service I get caller ID and on my faktortel service I get caller ID as well. I cant recommend the www.faktortel.com.au service enough. I pay $12 a month and can have a Sydney or Melbourne phone number that my Australian work colleagues can call me on for unlimited incoming calls at $12 a month. I'm a heavy user of this incoming service (I use my packet 8 for outgoing) and have had about 1 bad call a month out of 50 or so. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-8307-3503 (Sydney in-dial) > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Nathan E. Pralle > Sent: Thursday, 26 May 2005 11:06 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] International Caller ID? > > Greetings. > > My wife and I make a LOT of International calls -- mostly to Australia. I > just got Caller ID on my landline, but no numbers/names get passed on > international calls. Is it even possible to get this? Is it a special > request to the phone company, or is it just not possible at all? > > Thanks, > Nathan > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco Call Manager & Asterisk for Voicemail
While not widely used in the Cisco product line, SCCP can be used for more than handsets. Newer VoIP gateways support SCCP trunking. SIP in the 4.X series of CCM is a nice addition, but it is rather limited at the moment. G7.11 only, requires a MTP for DTMF, hold and transfer, etc. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Davidson Sent: Thursday, May 26, 2005 7:59 PM To: Scott Herrick Cc: Asterisk Users Mailing List - Non-Commercial Discussion; Shaun Ewing Subject: Re: [Asterisk-Users] Cisco Call Manager & Asterisk for Voicemail On 5/26/05, Scott Herrick <[EMAIL PROTECTED]> wrote: > My environment is a standard Cisco Call Mangler VoIP solution that has > reached the capacity of the Unity VM system. The cost of another Unity > box is enough to prompt the decision makers to look for other solutions. > > All the calls come into a Cisco 6509 and then to the CM. > > The more I look at the H.323 the less I like it. Rather, I didn't know > how much I didn't know. And I thought SIP was a thick protocol. > > I have not thought this through but would SCCP on the * box help? Might > there be a way to have * "talk" SCCP to CM for VM and MWI? > > Back to the books > Scott My environment could be a practical clone of yours. Truth be told, under CCM, h.323 is the best option out there, at least until you get to CCM 4.0. I share your trepidation with SIP- and it's not really an option under CCM 3- but SIP and H.323 share many of the same problems regarding NAT traversal, etc. H.323 is more of a logical descendant from ISDN, however, and as such is a little more robust- but it's still pretty cryptic. SCCP will not help at all. CCM uses SCCP only for communication with handsets, and to my knowledge, the protocol was not designed to handle server to server communication. MGCP would be an option, except it's possibly worse than h.323, and Asterisk doesn't yet support it in the way CCM would need to see Asterisk as an MGCP gateway. As an interesting aside, we just learned from Cisco that later generation FXO modules for the router VIC modules support Caller-ID back to Callmanager- but *ONLY* if you define the VIC as an h.323 gateway- MGCP doesn't support passing the fields back to CCM. Store that one away, those of you who are using CCM. MWI, on the other hand, is one area that I think Cisco got right, at least from an implementation standard under CCM. If you want to turn on the MWI, simply place a call to a extension of your choosing, while setting caller ID info for the call to be the extension you want to turn on- CCM does the rest. So, a simple call file will take care of the whole mess- off, on.. that's what Shaun spent the time putting together. It requires at least a signalling handshake between CCM and Asterisk- it actually doesn't need a voice channel setup from Asterisk to CCM, but of course, if you can do signalling, having voice is more than a bonus. The good news? You don't really have to know all that much about h.323 or sip to configure the trunks and get Asterisk and CCM humming to the same tune. You've already got a strong networking environment (6509s are nice iron), so connecting CCM and Asterisk really isn't much of a process- follow Shaun's example to the letter, possibly with my modification if it helps you, and you're golden- once you've got it configured, many other things will start to click. Good luck. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP PLZ$B!'(Bsip channel & AGI problem
Hi, ALL:
I use asterisk -r and "sip debug" to debug my sip channel.
And I build my sip proxy(5060) and asterisk(5065) on the same machine.
I make a call from 1011 to on sip proxy,
sip proxy forwards this call to "0939749001".
And this 0939749001 will be sent to asterisk by sip proxy.
sip ua(1011) => sipproxy => sip ua ( call forward 0939749001)
||
==> asterisk ===> cisco 5300 ==>
0939749001 (pstn)
I can find $EXTEN is equal to 0939749001 ( a mobile phone number )
and my $CALLERIDNUM is 1011
But how can I get the value of "" from "To:" field? ( via this sip ua)
In another word, I want to record the "middle" man.
My extensions.conf :
exten => _.,1,Answer
exten => _.,2,DeadAGI(my.agi,${CALLERIDNUM},${EXTEN})
exten => _.,3,Hangup
My log on asterisk CLI:
-- Executing DeadAGI("SIP/61.220.xxx.xxx-081888c8",
"my.agi|1011|0939749001|4") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
ser*CLI>
<-- SIP read from 61.220.xxx.xxx:5060:
ACK sip:[EMAIL PROTECTED]:5065 SIP/2.0
Record-Route:
Via: SIP/2.0/UDP 61.220.xxx.xxx;branch=0
Via: SIP/2.0/UDP
220.134.18.190:47286;rport=47286;branch=z9hG4bKB90B5F6F80294C48AACF7BDE31B9D2F1
From: 1011 ;tag=915860198
To: ;tag=as1c0a7e38<=== I want to get this value
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 57194 ACK
Max-Forwards: 16
Content-Length: 0
--
Best Regards
Charles
--
Best Regards
Charles
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[Asterisk-Users] International Caller ID?
We have antiquated caller ID schemes here in Australia. We barely support numbers from other local carriers, let alone OS ones. Certainly no names either. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analog Telephone Adapter
An IBM sales rep once told me... I can give you RELIABLE, FAST and CHEAP... any two of them at once. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog Telephone Adapter
I'll be trying AG-468 4 x FXS about 88.00USD from ATComm and let you know when I get one (though it might be a while) -- #Joseph On Thu, 2005-05-26 at 18:56 -0400, Waldo Rubinstein wrote: > I'm looking for a good, reliable, and cheap 4-port FXS ATA. Does > anyone know of one that works with Asterisk? > > Thanks, > Waldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] International Caller ID?
On Thu, 2005-05-26 at 22:05 -0500, Nathan E. Pralle wrote: > Greetings. > > My wife and I make a LOT of International calls -- mostly to Australia. I > just got Caller ID on my landline, but no numbers/names get passed on > international calls. Is it even possible to get this? Is it a special > request to the phone company, or is it just not possible at all? Most dont offer it on landlines. VoIP is a special case becuase of how its done. Name is also gonna be a problem because of how that is looked up, odds are the telco wont be able to provide that even if they can provide the number. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] International Caller ID?
Greetings. My wife and I make a LOT of International calls -- mostly to Australia. I just got Caller ID on my landline, but no numbers/names get passed on international calls. Is it even possible to get this? Is it a special request to the phone company, or is it just not possible at all? Thanks, Nathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Call Manager & Asterisk for Voicemail
On 5/26/05, Scott Herrick <[EMAIL PROTECTED]> wrote: > My environment is a standard Cisco Call Mangler VoIP solution that has > reached the capacity of the Unity VM system. The cost of another Unity > box is enough to prompt the decision makers to look for other solutions. > > All the calls come into a Cisco 6509 and then to the CM. > > The more I look at the H.323 the less I like it. Rather, I didn't know > how much I didn't know. And I thought SIP was a thick protocol. > > I have not thought this through but would SCCP on the * box help? Might > there be a way to have * "talk" SCCP to CM for VM and MWI? > > Back to the books > Scott My environment could be a practical clone of yours. Truth be told, under CCM, h.323 is the best option out there, at least until you get to CCM 4.0. I share your trepidation with SIP- and it's not really an option under CCM 3- but SIP and H.323 share many of the same problems regarding NAT traversal, etc. H.323 is more of a logical descendant from ISDN, however, and as such is a little more robust- but it's still pretty cryptic. SCCP will not help at all. CCM uses SCCP only for communication with handsets, and to my knowledge, the protocol was not designed to handle server to server communication. MGCP would be an option, except it's possibly worse than h.323, and Asterisk doesn't yet support it in the way CCM would need to see Asterisk as an MGCP gateway. As an interesting aside, we just learned from Cisco that later generation FXO modules for the router VIC modules support Caller-ID back to Callmanager- but *ONLY* if you define the VIC as an h.323 gateway- MGCP doesn't support passing the fields back to CCM. Store that one away, those of you who are using CCM. MWI, on the other hand, is one area that I think Cisco got right, at least from an implementation standard under CCM. If you want to turn on the MWI, simply place a call to a extension of your choosing, while setting caller ID info for the call to be the extension you want to turn on- CCM does the rest. So, a simple call file will take care of the whole mess- off, on.. that's what Shaun spent the time putting together. It requires at least a signalling handshake between CCM and Asterisk- it actually doesn't need a voice channel setup from Asterisk to CCM, but of course, if you can do signalling, having voice is more than a bonus. The good news? You don't really have to know all that much about h.323 or sip to configure the trunks and get Asterisk and CCM humming to the same tune. You've already got a strong networking environment (6509s are nice iron), so connecting CCM and Asterisk really isn't much of a process- follow Shaun's example to the letter, possibly with my modification if it helps you, and you're golden- once you've got it configured, many other things will start to click. Good luck. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Call Manager & Asterisk for Voicemail
My environment is a standard Cisco Call Mangler VoIP solution that has
reached the capacity of the Unity VM system. The cost of another Unity
box is enough to prompt the decision makers to look for other solutions.
All the calls come into a Cisco 6509 and then to the CM.
The more I look at the H.323 the less I like it. Rather, I didn't know
how much I didn't know. And I thought SIP was a thick protocol.
I have not thought this through but would SCCP on the * box help? Might
there be a way to have * "talk" SCCP to CM for VM and MWI?
Back to the books
Scott
Paul Davidson wrote:
In my particular environment, transfer directly into VM is not
important- we're geographically diverse, with almost no administrative
support- so you either answer your phone, or you forward your phone to
voicemail and let the voicemail do the work.
10 minutes with the debugger just yielded some very interesting
results, if you want to have all VM use a single number (so you don't
have to set up a separate extension for each VM box, and don't have to
do more than check 'on busy, forward to voicemail' on each extension.
It only requires a slight modification to your instructions, can run
in parallel with your solution, and should be simple to test. The
only drawback is that you lose the caller-id info of the original
caller. Here's what you do:
Set up a new trunk, precisely as you've configured your existing
trunk- again, make sure it's two way, and verify that it works
properly. To do this verification, set up a new route pattern, and
point it to your new trunk- calling the new route pattern should work
precisely as your other ones do.
Make one small modification to your new trunk. Under the 'Outbound
Calls' section, 'Calling Party Selection', set it to 'First Redirect
Number'. Yours is most likely set to 'Originator'.
Change your dialplan to add a new 'all voicemail' extension, that
matches the number on your new Route Pattern- there should be no need
for XXX's, just straight numbers all the way through- you want an
exact match every time. The dialplan (assuming your new extension is
1799, is as follows:
exten => 1799,1,NoOp
exten => 1799,2,MailBoxExists([EMAIL PROTECTED])
exten => 1799,3,Congestion
exten => 1799,103,Voicemail(su${CALLERID})
exten => 1799,104,Playback(vm-goodbye)
exten => 1799,105,Hangup
As I said- the caller's caller id info is lost in the process- Cisco
doesn't seem to send any redirection fields in the PDUs that I can
find, so you have to play with the caller info- but the rest should
work like a charm.
-pbd
On 5/26/05, Shaun Ewing <[EMAIL PROTECTED]> wrote:
On 5/26/05, Paul Davidson <[EMAIL PROTECTED]> wrote:
You've done the hard bits.
The bad news is that, under CCM, there's really not much in the way of
VM configuration. You should set up the VM Pilot stuff to your
extension for the Asterisk voicemail- this allows you to click the
'voicemail' box on each extension rather than keying it in- but you
still have to touch each extension. You can use their automated tools
to make systemwide changes to all extensions- but I don't trust them
at all, and I don't think that would help you in this case.
Yep, I've setup a VM Pilot. I changed the default pilot, so the
messages key works on all phones.
Phones without a mailbox, Asterisk prompts for mailbox and password.
Phones with a mailbox, just the password.
I'd love to see how you configured the MWI and how you've set your
dialplan- from the way it looks, you're using a different extension
for each mailbox. Theoretically, there should be fields on the PDUs
from h.323 that show the forwarding number- that's the way Unity does
it- and you go into VM for the forwarding number, not for the
extension dialed. I'm not sure without playing if any of the h323
channel drivers make the forwarding number available as a channel
variable- if they don't, it should be a relatively trivial patch,
assuming CCM sends it across (which I'm pretty sure it does- again,
time to set some debugs and watch the PDUs).
The notes basically show how MWI is configured.
I am actually using a different extension for each mailbox. This is
something I setup a while ago to allow calls to be transferred direct
to somebody's mailbox, and it has proven useful for this as well.
-pbd
-Shaun
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Re: [Asterisk-Users] What does Asterisk need in the way of a GUI?
Please do yourself and everyone involved a favor and use an open and widely used management standard such as WBEM. Start at www.openwbem.org and/or http://sblim.sourceforge.net/ for a jumpstart on WBEM. I had started on this already, but got sidetracked on other projects and never got back to it. I'll be happy to answer any questions you might have on the subject, to the best of my ability. Greg Mitchel Constantin wrote: We are two programmers who are passionate for Asterisk and we will be dedicating the next three months towards programming for Asterisk and would like to get some input from everyone on what they feel Asterisk is lacking or needs based on what is not currently a part of it or available through third parties. Hopefully, by asking up front we won't be wasting our time on something nobody wants or needs. Specifically I am asking in the way of GUI's (web-based or not), not in backend programming as Mark and others have that well under control! Thank you for your suggestions, Mitchel & Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What does Asterisk need in the way of a GUI?
Do you need volunteers? My employer has just given the go ahead to devote my time to a project like this Sean On Thu, 2005-05-26 at 18:34 -0700, Mitchel Constantin wrote: > We've collaborated, and are going to work on an advanced GUI client > with a web interface to compliment it, it will be an all in one type > of system. > > I would really appreciate feature requests on what you would like in a > windows/linux form client. Please be creative =). We already have a > very feature rich list planned! > > Thank you, > Mitchel > > On 5/25/05, admin <[EMAIL PROTECTED]> wrote: > > > > > > Here are a couple of items I hear people asking for regularly. > > > > - Multi-tenant functionality > > - Allow users to change their own preferences via web (call forwarding, MoH, > > > > etc...) > > > > > > > We are two programmers who are passionate for Asterisk and we will be > > > dedicating the next three months towards programming for Asterisk and > > > would like to get some input from everyone on what they feel Asterisk > > > is lacking or needs based on what is not currently a part of it or > > > available through third parties. Hopefully, by asking up front we > > > won't be wasting our time on something nobody wants or needs. > > > > > > Specifically I am asking in the way of GUI's (web-based or not), not > > > in backend programming as Mark and others have that well under > > > control! > > > > > > Thank you for your suggestions, > > > Mitchel & Tom > > > ___ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AreskiCC
I'm tring to dowload the AreskiCC but the www.areski.net is out of order, There are any other place that I can download it ? Miguel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PHP/AGI Problem
On Thu, 2005-26-05 at 21:23 -0400, Michael Stearne wrote: > Thanks for the efforts Matt. David Eder is the one who deserves the praise, he's done wonderful work on version 2. Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What does Asterisk need in the way of a GUI?
We've collaborated, and are going to work on an advanced GUI client with a web interface to compliment it, it will be an all in one type of system. I would really appreciate feature requests on what you would like in a windows/linux form client. Please be creative =). We already have a very feature rich list planned! Thank you, Mitchel On 5/25/05, admin <[EMAIL PROTECTED]> wrote: > > > Here are a couple of items I hear people asking for regularly. > > - Multi-tenant functionality > - Allow users to change their own preferences via web (call forwarding, MoH, > > etc...) > > > > We are two programmers who are passionate for Asterisk and we will be > > dedicating the next three months towards programming for Asterisk and > > would like to get some input from everyone on what they feel Asterisk > > is lacking or needs based on what is not currently a part of it or > > available through third parties. Hopefully, by asking up front we > > won't be wasting our time on something nobody wants or needs. > > > > Specifically I am asking in the way of GUI's (web-based or not), not > > in backend programming as Mark and others have that well under > > control! > > > > Thank you for your suggestions, > > Mitchel & Tom > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoiPSupply Dot Com
Karl, first off I apologize for any inconvenience on your recent order. I will take a look at your transaction to see where things may have gone awry. We do make mistakes, but we strive to not make the same mistake more than once. Secondly, I apologize to the list moderator for the pseudo-commercial nature of this post. The grievance was aired on this list, and I felt compelled to respond to this list and I realize much of this may be more appropriate for the BIZ list. With respect to Cisco phone licensing, I myself sometimes become confused. Cisco has retooled this program a few times. Different resellers have access to various licensed versions, including CallManager (CH1) ,CallManager Express (CCME) and the newer SIP licensed units. We turn over stock on Cisco phones very rapidly and source stock from all over the globe. If we happen to have a large stock position of phones that include a particular type of license, we try to be proactive in updating the individual product page to reflect the version of the phones we are currently selling. So you are not disappointed in the future, unless we are specifically advertising that the phones include a specific version license, best to assume you are going to get the Spare (unlicensed) version. To my knowledge, none of the Cisco models come with SIP preloaded, as Cisco's vested interest lies in SCCP the native protocol of their CallManager platform. Better yet, call one of our inside sales reps and ask us to ship or source the specific version of phone you require for your application. If you get someone named "Damon", hang up, check to make sure you are dialing the right number, and call again. We have a lot of new hires, and I have trouble keeping track of all the new faces, but I am certain there is no one in our employ by the name of Damon. We do have a Ramon and a Darren. If you can give me a positive id here I will see to it they are fired by 9:05AM tomorrow. Just kidding, That being said, due to the growth of the VOIP industry, and the resultant growth of our company and customer base, we spend a great deal of time retooling our infrastructure and instituting the necessary manpower, policies and procedures to efficiently and correctly process larger volumes of customer orders. It is a challenge to orchestrate the logistics sometimes but I can assure you we want our customers to value our service. It is our policy to only ship credit card orders to an address that is on file with with the card issuing bank. If we relied solely on our merchant clearing house to authorize credit card transactions, we would be out of business in short order. We receive countless fraudulent transactions, and therefore we must manually verify the identity of the card holder in many cases. If the ship to: address is not on file with the card issuing bank, we quickly contact the customer and ask them to call their issuing bank and get the alternate address on file. Once that has been completed, the order is released to shipping. If your order fits this scenario, that may have accounted for the unfortunate, but necessary, delay. Many of our customers have had to endure some lengthy backorder/delays on popular products because of supply issues with our vendors. We have great relationships with our vendors and I spend most of my day on the phone performing minor miracles in an effort to keep the supply chain moving. We appreciate the kind words from what appear to be the majority of respondents to Karl's post. To any and all who feel we may have dropped the ball, we hope you'll give us a second look. In appreciation for the positive reinforcement from our customers on the list, and with respect to Karl's grievance and constructive criticism, I have created a discount code which will remain active until the end of the month. If anyone reading this has equipment needs, and wants to save 10% , simply enter the discount code SORRYKARL in the appropriate field during the checkout procedure. Like I said, this code will be valid until the end of the month, good for WEB ORDERS ONLY (please don't ask your sales rep to honor the discount if you are phoning in your order, these guys rely on commissions for a portion of their income). The discount code is case sensitive so make sure you lock the caps. I wish you all continued success in your telephony endeavors! Cory AndrewsPartner / PurchasingVOIPSupply.com++454 Sonwil DriveBuffalo, NY 14225++v - 800.398.VOIP Ext 22f - 716.630.1548e - [EMAIL PROTECTED] -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Karl J. VesterlingSent: Thursday, May 26, 2005 6:08 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] VoiPSupply Dot ComAbout this business with the distraction of the Cisco licenses...
Re: [Asterisk-Users] PHP/AGI Problem
On 5/26/05, Matthew Asham <[EMAIL PROTECTED]> wrote: > On Thu, 2005-26-05 at 17:59 -0400, Michael Stearne wrote: > > On 5/26/05, Benjamin West <[EMAIL PROTECTED]> wrote: > > > Michael, > > > The version, in the context of Jon's problem, was irrelevant. Jon's > > > problem was due to a small bug in his code, and not related to PHPAGI. > > > > Yeah. I was just wondering what version people were using since 2.0 > > hasn't been formally released yet. I spoke to the developer and he > > suggested 2.0. > > > > phpagi-2.14 has now been released, this is the first "official" release > for the 2.x series but should be stable. > > There is also a compatibility module in CVS to support older 1.x apps. > > You can snag it from http://phpagi.sourceforge.net/ Thanks for the efforts Matt. Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tds_CDR and MS SQL Server troubleshooting
did you follow the directions in doc/cdr.txt in the asterisk distro? Duane Cox - Original Message - From: "PA" <[EMAIL PROTECTED]> To: Sent: Thursday, May 26, 2005 3:16 PM Subject: [Asterisk-Users] tds_CDR and MS SQL Server troubleshooting I am trying to get * to write CDR records to an MS SQL table. I am used the freeTDS set up (not using UNIX ODBC), and it appears to be making the connection to the database, but I make a call, I get this error: WARNING[3369]: cdr_tds.c:180 tds_log: Reconnected to SQL database. ERROR[3369]: cdr_tds.c:191 tds_log: Failed to insert Call Data Records into SQL database. I've assigned all rights to the cdr table in SQL to the user. I am not sure how to troubleshoot this from this point. Does anyone know else these errors are? Thanks for any direction. Phil ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pressing a key to get in of voicemail?
I've currently got Asterisk configured to take incoming calls, ask for extension, ring the phone and send them directly to the voicemail. What I want to be able to do is first a message "press 1 for voicemail or hangup" before voicemail come up. Any ideas? regards, -- Ing CIP Alejandro Celi Mariátegui <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CRM integration (was RE: CallerID)
Thx Michiel, Ill grab it again and examine... Thx! |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Michiel van Baak |Sent: Jueves, 26 de Mayo de 2005 02:22 p.m. |To: asterisk-users@lists.digium.com |Subject: Re: [Asterisk-Users] CRM integration (was RE: CallerID) | |On 18:48, Wed 25 May 05, Anton Krall wrote: |> Michiel. |> |> To help better understand what xmlhttp does with asterisk, could you |> modify your tgx file and strip everything related to db and make a |> sort of hello world script? In fact, something small like connecting |> to the manager (of course, we each have to put our config there) and |> maybe issue a sip show peers and show how this can be done |and update in realtime via web? |> |> What do you think? This will elp grasp the concept of xmlhttp and |> connecting to the manager. |> | |Anton, | |My script is not connecting to the manager interface. |The php script is run as agi script as first when a call comes |in. The php script will know the callerID of the calling |party, that is something the AGI interface provides to the script. |Then the AGI script connects to my CRM database to do a lookup |on the phone number. If it finds a match it will put the |companyName and the records ID in a temporary table in the |same database. Of course instead of storing it in a database |it can also write a little xml file. I think I will do that in |the next release of our CRM app (Where this code is taken from). |I altered the agi script to simplify it a little and I added |comments to explain what is being done. |I will also do this to the php script that generates the |callerID notification in my CRM app. |I will upload the new agi to my server as soon as this mail is |sent. Just for the record: |http://michiel.vanbaak.info/Files/ | |Have fun, | |> |> |> |-Original Message- |> |From: [EMAIL PROTECTED] |> |[mailto:[EMAIL PROTECTED] On Behalf |Of Michiel |> |van Baak |> |Sent: Mi?rcoles, 25 de Mayo de 2005 02:04 p.m. |> |To: asterisk-users@lists.digium.com |> |Subject: Re: [Asterisk-Users] CRM integration (was RE: CallerID) |> | |> |On 10:17, Wed 25 May 05, Rusty Shackleford wrote: |> |> > -Original Message- |> |> > From: [EMAIL PROTECTED] |> |> > [mailto:[EMAIL PROTECTED] On Behalf |> |Of Anton |> |> > Krall |> |> > Sent: Wednesday, May 25, 2005 7:41 AM |> |> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |> |> > Subject: RE: [Asterisk-Users] CRM integration (was RE: CallerID) |> |> > |> |> > |> |> > It doesn?t seem to be complicated but for example, the things |> |> > that bother me are refreshes, I don?t want to use meta |> |refreshes for this |> |> > monitoring webpage every X seconds, rather, use something more |> |> > realtime... Any ideas? |> |> |> |> And that's the real trick. Web browsers, unless they are |> |instructed to |> |> do otherwise, don't DO anything once they've completed |> |loading a page. |> |> So without instructing them to refresh, they aren't going to |> |be aware |> |> of a server-side change, such as an incoming call. For |that, you're |> |> going |> | |> |This is not true. If it was for pure HTML only, yes, you |are correct. |> |But with javascript you can start a timer and execute a javascript |> |function every once in a while. |> |If this javascript loads an XML document off the server, |you're there |> |;) |> | |> | |> |> to have to have some way of sending a message TO the client |> |> machine, have it received by that machine, and have that client |> |> machine take the desired action (pop up an incoming call dialog, |> |> load a contact record, etc.). |> |> |> | |> |Have a look at the tgz file I posted earlier |> | |> |> -- |> |> No virus found in this outgoing message. |> |> Checked by AVG Anti-Virus. |> |> Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: |> |> 05/20/2005 |> |> |> |> |> |> ___ |> |> Asterisk-Users mailing list |> |> Asterisk-Users@lists.digium.com |> |> http://lists.digium.com/mailman/listinfo/asterisk-users |> |> To UNSUBSCRIBE or update options visit: |> |>http://lists.digium.com/mailman/listinfo/asterisk-users |> | |> |-- |> |Michiel van Baak |> |http://lunteren.vanbaak.info |> |[EMAIL PROTECTED] |> |GnuPG key: |> |http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D |> | |> |"Two of the most famous products of Berkeley are LSD and BSD. |> |I don't think that this is a coincidence." |> | |> |___ |> |Asterisk-Users mailing list |> |Asterisk-Users@lists.digium.com |> |http://lists.digium.com/mailman/listinfo/asterisk-users |> |To UNSUBSCRIBE or update options visit: |> | http://lists.digium.com/mailman/listinfo/asterisk-users |> | |> |> ___ |> Asterisk-Users mailing list |> Asterisk-Users@lists.digium.com |> http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial to Ineen
Colin Anderson wrote:
This looks pretty interesting:
www.ineen.com
Rebranded Eyebeam softphone.
You can call an ineen user with this in your dialplan:
exten => _5.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
I tried that, but it gave me
-- Executing Dial("SIP/615-811a", "SIP/[EMAIL PROTECTED]|60|tr") in
new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 488 "Not Acceptable Here" back from 64.34.98.148
== No one is available to answer at this time (1:0/0/0)
May 27 08:43:09 NOTICE[9733]: rtp.c:453 ast_rtp_read: RTP: Received
packet with bad UDP checksum
-- Timeout on SIP/615-811a
Any ideas?
bye
Ronald
I tried to call my own Asterisk server by using [EMAIL PROTECTED] syntax and
I got 401/Unauthorized which is exactly what I expected to get. I added the
ineen account to sip.conf and I got 407/Proxy Authentication Required.
h. Password problem, eh? I'll fire up Ethereal and see what's going on
there.
Haven't tried the video yet. Anyone else tried this thing?
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RE: [Asterisk-Users] Looking for inexpensive phone to use with Asteriskwith message light and a button that will let me play new messages
How about "almost" -- I got three of these: http://www.sciplus.com/singleItem.cfm?terms=4264&cartLogFrom=Search%20%2 D%20Category%20Filter&CFID=8246070&CFTOKEN=33690221 Each comes with two red LEDs installed. I left one unit red, changed one to blue and another to green, then installed them in the kitchen near our message board. Each is connected to a different phone-line (from SPA-2000). Now we have "his 'n hers" message indicators and one for my business line. > -Original Message- > From: Kellner, Peter [mailto:[EMAIL PROTECTED] > Sent: Thursday, May 26, 2005 2:04 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Looking for inexpensive phone to > use with Asteriskwith message light and a button that will > let me play new messages > > > I'm wanting to have a phone at home next to the garage door > that when my bride comes home, she can see that there is a > new message, push a button and have the messages played to > her. Otherwise, she will not let me install asterisk on my home line. > > Can someone suggest relatively inexpensive hardware that will > do this for me (us)? > > Thanks, > > -Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PHP/AGI Problem
On Thu, 2005-26-05 at 17:59 -0400, Michael Stearne wrote: > On 5/26/05, Benjamin West <[EMAIL PROTECTED]> wrote: > > Michael, > > The version, in the context of Jon's problem, was irrelevant. Jon's > > problem was due to a small bug in his code, and not related to PHPAGI. > > Yeah. I was just wondering what version people were using since 2.0 > hasn't been formally released yet. I spoke to the developer and he > suggested 2.0. > phpagi-2.14 has now been released, this is the first "official" release for the 2.x series but should be stable. There is also a compatibility module in CVS to support older 1.x apps. You can snag it from http://phpagi.sourceforge.net/ -- Matthew Asham - the B.C. Wireless Network Society www.bcwireless.net - +1 604 484 5289 x1006 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Sound List in HTML - Updated
Greetings all. Well, the first Asterisk Sound List in HTML was so popular, I did some more fiddling around to make it even more useful. Here's an updated page: - One master list with all sounds, sorted alphabetically by filename - The old lists are linked from the master list - Shows what directory the sound is in - Shows what package it comes with (default, extras) - Shows the length of the sound in seconds, to the hundredth of a second - The file name is a link to an the MP3 file of the sound so you can easily listen to the sound before using it inside Asterisk. http://www.nathanpralle.com/software/ast_masterlist.html Comments/suggestions appreciated! Enjoy! Nathan Pralle ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM400P in 2U server?
I'm curious about why they bother asking for low profile ram in something like this, 2RU should have plenty of clearance for average ram. Dean > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Emile Bourquin > Sent: Thursday, 26 May 2005 7:40 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] TDM400P in 2U server? > > Hello, this is a question for the Asterisk systems-integrators; I am > creating a rackmount Asterisk server, and would like to start with a > TDM400P > card. I believe the TDM400P card would work in this server, with the PCI-X > configuration: > > http://www.aberdeeninc.com/abcatg/MB1130.htm , which is the this: > http://supermicro.com/products/system/2U/6024/SYS-6024H-82R.cfm > > Is this correct? I'm concerend about the height of the card, and whther it > will work in a PCI-X slot. Thanks for any help! > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limiting maximum runtime of echo test
Johnathan Corgan schrieb: Bastian Schern wrote: is it possible to limit the maximum runtime of the command "echo"? Use the AbsoluteTimeout application in your dialplan preceding the Echo application. http://voip-info.org/tiki-index.php?page=Asterisk%20AbsoluteTimeout -Johnathan Thanks allot Bastian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P in 2U server?
Hello, this is a question for the Asterisk systems-integrators; I am creating a rackmount Asterisk server, and would like to start with a TDM400P card. I believe the TDM400P card would work in this server, with the PCI-X configuration: http://www.aberdeeninc.com/abcatg/MB1130.htm , which is the this: http://supermicro.com/products/system/2U/6024/SYS-6024H-82R.cfm Is this correct? I'm concerend about the height of the card, and whther it will work in a PCI-X slot. Thanks for any help! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiPSupply Dot Com
It's not the licenses, that's like 10% of the problem. One can always buy licenses... But the 16 man hours I wasted waiting for the across-town overnight shipment vastly outweighed the cost of the licenses. To mitigate risk, why didn't you ask to pick up the product in person? The main gripe is: It's the fact that the next-day delivery for across town was two days late. Add to that the fact that it was shipped to the Bill-To address and not the Ship-To address. Thereby causing me to bill out 16 hours of my time (which isn't cheap), for sitting on my hands. THAT IS THE PROBLEM! I don't have any feelings about Voipsupply.com one way or another, but you're really beating a dead horse here about something that should be common sense. Like Ronald Reagan said about the Soviets: "Trust But Verify." Translation: get a tracking confirmation number before you travel and waste your valuable time next time. Sh*t happens with shipping no matter how diligent the supplier is. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Call Manager & Asterisk for Voicemail
In my particular environment, transfer directly into VM is not
important- we're geographically diverse, with almost no administrative
support- so you either answer your phone, or you forward your phone to
voicemail and let the voicemail do the work.
10 minutes with the debugger just yielded some very interesting
results, if you want to have all VM use a single number (so you don't
have to set up a separate extension for each VM box, and don't have to
do more than check 'on busy, forward to voicemail' on each extension.
It only requires a slight modification to your instructions, can run
in parallel with your solution, and should be simple to test. The
only drawback is that you lose the caller-id info of the original
caller. Here's what you do:
Set up a new trunk, precisely as you've configured your existing
trunk- again, make sure it's two way, and verify that it works
properly. To do this verification, set up a new route pattern, and
point it to your new trunk- calling the new route pattern should work
precisely as your other ones do.
Make one small modification to your new trunk. Under the 'Outbound
Calls' section, 'Calling Party Selection', set it to 'First Redirect
Number'. Yours is most likely set to 'Originator'.
Change your dialplan to add a new 'all voicemail' extension, that
matches the number on your new Route Pattern- there should be no need
for XXX's, just straight numbers all the way through- you want an
exact match every time. The dialplan (assuming your new extension is
1799, is as follows:
exten => 1799,1,NoOp
exten => 1799,2,MailBoxExists([EMAIL PROTECTED])
exten => 1799,3,Congestion
exten => 1799,103,Voicemail(su${CALLERID})
exten => 1799,104,Playback(vm-goodbye)
exten => 1799,105,Hangup
As I said- the caller's caller id info is lost in the process- Cisco
doesn't seem to send any redirection fields in the PDUs that I can
find, so you have to play with the caller info- but the rest should
work like a charm.
-pbd
On 5/26/05, Shaun Ewing <[EMAIL PROTECTED]> wrote:
> On 5/26/05, Paul Davidson <[EMAIL PROTECTED]> wrote:
> >
> > You've done the hard bits.
> >
> > The bad news is that, under CCM, there's really not much in the way of
> > VM configuration. You should set up the VM Pilot stuff to your
> > extension for the Asterisk voicemail- this allows you to click the
> > 'voicemail' box on each extension rather than keying it in- but you
> > still have to touch each extension. You can use their automated tools
> > to make systemwide changes to all extensions- but I don't trust them
> > at all, and I don't think that would help you in this case.
>
> Yep, I've setup a VM Pilot. I changed the default pilot, so the
> messages key works on all phones.
>
> Phones without a mailbox, Asterisk prompts for mailbox and password.
> Phones with a mailbox, just the password.
>
> > I'd love to see how you configured the MWI and how you've set your
> > dialplan- from the way it looks, you're using a different extension
> > for each mailbox. Theoretically, there should be fields on the PDUs
> > from h.323 that show the forwarding number- that's the way Unity does
> > it- and you go into VM for the forwarding number, not for the
> > extension dialed. I'm not sure without playing if any of the h323
> > channel drivers make the forwarding number available as a channel
> > variable- if they don't, it should be a relatively trivial patch,
> > assuming CCM sends it across (which I'm pretty sure it does- again,
> > time to set some debugs and watch the PDUs).
>
> The notes basically show how MWI is configured.
>
> I am actually using a different extension for each mailbox. This is
> something I setup a while ago to allow calls to be transferred direct
> to somebody's mailbox, and it has proven useful for this as well.
>
> > -pbd
>
> -Shaun
>
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Re: [Asterisk-Users] TDM400P vs SIP3000 x2
I just got through trying to set up a Sipura 3000 and am still looking for answers. There is a low volume problem (caller is underwater) on the FXO port that I wish someone would have told me about and I would have gone the other route. (even after upgrading firmware and adjusting gain settings) More details here. http://voxilla.com/index.php?name=PNphpBB2&file=viewtopic&t=3500&highlight=vol+volume http://voxilla.com/index.php?name=PNphpBB2&file=viewtopic&t=1249&highlight=vol+volume Maybe they work great and this one is defective, but others appear to have a simular problem and this was my experience. Dan On 5/26/05, Brian Roy <[EMAIL PROTECTED]> wrote: > On 5/26/05, Andres Paglayan <[EMAIL PROTECTED]> wrote: > > > > > I am about to start building my first ever * production server and would > > be nice to have some input from the list. > > My personal vote would be for the Sipura's. > > Pro's - > > It would make failing over to standby box much easier. > You could run a small 1u box and not have to worry about PCI requirements. > Lightens the load (especially interrupts) on the * box > PSTN doesn't have to be located by the * box, just by an ethernet port > > I think if you poll the archives, you would find problems with both of > them. I run a SPA3k and have had no problems with it at all. > > Just my .02 > > -Chuji > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analog Telephone Adapter
I'm looking for a good, reliable, and cheap 4-port FXS ATA. Does anyone know of one that works with Asterisk? Thanks, Waldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Call Manager & Asterisk for Voicemail
On 5/26/05, Paul Davidson <[EMAIL PROTECTED]> wrote: > > You've done the hard bits. > > The bad news is that, under CCM, there's really not much in the way of > VM configuration. You should set up the VM Pilot stuff to your > extension for the Asterisk voicemail- this allows you to click the > 'voicemail' box on each extension rather than keying it in- but you > still have to touch each extension. You can use their automated tools > to make systemwide changes to all extensions- but I don't trust them > at all, and I don't think that would help you in this case. Yep, I've setup a VM Pilot. I changed the default pilot, so the messages key works on all phones. Phones without a mailbox, Asterisk prompts for mailbox and password. Phones with a mailbox, just the password. > I'd love to see how you configured the MWI and how you've set your > dialplan- from the way it looks, you're using a different extension > for each mailbox. Theoretically, there should be fields on the PDUs > from h.323 that show the forwarding number- that's the way Unity does > it- and you go into VM for the forwarding number, not for the > extension dialed. I'm not sure without playing if any of the h323 > channel drivers make the forwarding number available as a channel > variable- if they don't, it should be a relatively trivial patch, > assuming CCM sends it across (which I'm pretty sure it does- again, > time to set some debugs and watch the PDUs). The notes basically show how MWI is configured. I am actually using a different extension for each mailbox. This is something I setup a while ago to allow calls to be transferred direct to somebody's mailbox, and it has proven useful for this as well. > -pbd -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_misdn problem
Can you post the output of your asterisk log file and your initd script for starting mISDN. What versions of chan_misdn, ,mISDN and mISDNuser are you using. Also check to see that /dev/mISDN exists. Dave. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of me me Sent: Thursday, 26 May 2005 9:18 PM To: Asterisk Users Mailing List Subject: [Asterisk-Users] chan_misdn problem I've installed asterisk 1.0.7 with linux kernel 2.6.3 (patched for mISDN). I Compile mISDNuser and loaded de modules (hfcmulti, mISDNdsp) for my BN8S0 beronet card. I have installed chan_misdn-beta-0.0.3rc4 with no problems. I have configured my misdn.conf as follows: [general] context=default language=de debug=0 immediate=no hold_allowed=yes [octoBRI] ports=1,8,2,7,3,6,4,5 context=incoming msns=* when I start asterisk with asterisk -vvvc I get the following message and then asterisk dies: [chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found == Registered channel type 'mISDN' (This driver enables the asterisk to use hardware which is supported by the new ) cannot request MGR_NEWENTITY from mISDN: Success Ouch ... error while writing audio data: : Broken pipe Warning, flexible rate not heavily tested! Can anyone help me?? Thanks. __ Renovamos el Correo Yahoo! Nuevos servicios, más seguridad http://correo.yahoo.es ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.17 - Release Date: 25/05/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.17 - Release Date: 25/05/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Call Manager & Asterisk for Voicemail
Not to discourage you- but getting CCM and * to play well together with H.323 is beastly. MWI, following Shaun's instructions, *SHOULD* work just fine- I suspect you're running into the issue I had for a long time- CCM calls to * work fine- but * can't call CCM. I did eventually make it work- but only with chan_h323, and CVS HEAD- no stable branch works yet. I believe 1.2 will solve this- but don't even bother before CVS HEAD of about a month ago. chan_oh323 worked marginally better- but still didn't give me a functional way to call back from * to CCM. I think there's a basic fear of chan_h323 out there- lots of people trying asterisk-oh323 on the perception that it's somehow simpler- I've found absolutely no difference in pain configuring it from chan_h323- and the philosophy behind chan_h323 from a programmatic standpoint, I believe, is stronger (I'm not trying to start a war here, this is MY perception- I won't yell at you if you disagree with me). Frankly, the new h323 channel driver makes no sense to me- chan_h323 works perfectly well, I don't see any reason to invest time and effort to make a new one work- there's nothing superior about the technology whatsoever, and right now the feature set is anemic- completely useless to me in a CCM integration environment. I think it was summed up best in a note on mantis- there's now three channel drivers for one protocol- can someone please explain which one will be 'the one' going forward so I know where to place my resources? h323 must be popular- sip and iax only have one active channel development project going, sccp has two (three if you count the old branch of sccp vs sccp2)- that makes h.323 the number one thing going, right? Just my two cents, off topic as they may be. Kudos to Shaun for his paper- I'm going to take a couple of hours tonight seeing if CCM is sending over the forwarding number in the PDU. -pbd On 5/26/05, Scott Herrick <[EMAIL PROTECTED]> wrote: > Well, I have my CM and my Cisco phones leaving and retrieving VM from *. > :-) > The challenge now is the MWI. Shaun's notes are great! I discovered > the version of H323 is not playing well with asterisk-oh323. It looks > like I'm going to need to gut the h323 config on my * box and start with > a set of known working (interoperable) versions. > > It looks like if the MWI are working then the * VM call handlers/auto > attendants will also work. Has anyone got * doing Auto Attendants for a > Call Manager? > > I'm going to read a few tutorials on H.323 and OH.323. Back soon. > > Thanks > Scott > > Paul Davidson wrote: > > >>Date: Thu, 26 May 2005 16:50:02 +1000 > >>From: Shaun Ewing <[EMAIL PROTECTED]> > >>Subject: Re: [Asterisk-Users] Cisco Call Manager & Asterisk for > >>Voicemail > >>To: Asterisk Users Mailing List - Non-Commercial Discussion > >> > >>Message-ID: <[EMAIL PROTECTED]> > >>Content-Type: text/plain; charset=ISO-8859-1 > >> > >>On 5/26/05, Scott Herrick <[EMAIL PROTECTED]> wrote: > >> > >>>BUMP > >>>It's CM 3.3.6 > >>> > >>>MAN that would be sweet if * could take the place of Unity! > >>> > >>>Anybody? > >>> > >>>:-) > >> > >>I've got it working with Callmanager 3.3(5) and Asterisk (connected > >>with chan_oh323). > >> > >>Not totally integrated - one still needs to set call forwarding > >>(busy/no answer) on each extension that needs voicemail, but MWI works > >>and so does the messages button (eg: on 7960G) to retrieve VM. > >> > >>If somebody can tell me how to send a call in Callmanager to (for > >>example) extension 27000 when 7000 is unavailable by checking the > >>voicemail box (rather than entering an individual number for each > >>extension), it'll be perfect. > >> > >>I can share my progress so far if it will be beneficial. > >> > >>-Shaun > > > > > > You've done the hard bits. > > > > The bad news is that, under CCM, there's really not much in the way of > > VM configuration. You should set up the VM Pilot stuff to your > > extension for the Asterisk voicemail- this allows you to click the > > 'voicemail' box on each extension rather than keying it in- but you > > still have to touch each extension. You can use their automated tools > > to make systemwide changes to all extensions- but I don't trust them > > at all, and I don't think that would help you in this case. > > > > I'd love to see how you configured the MWI and how you've set your > > dialplan- from the way it looks, you're using a different extension > > for each mailbox. Theoretically, there should be fields on the PDUs > > from h.323 that show the forwarding number- that's the way Unity does > > it- and you go into VM for the forwarding number, not for the > > extension dialed. I'm not sure without playing if any of the h323 > > channel drivers make the forwarding number available as a channel > > variable- if they don't, it should be a relatively trivial patch, > > assuming CCM sends it across (which I'm pretty sure it does- again, > > time to set some debugs and
Re: [Asterisk-Users] Limiting maximum runtime of echo test
Bastian Schern wrote: is it possible to limit the maximum runtime of the command "echo"? Use the AbsoluteTimeout application in your dialplan preceding the Echo application. http://voip-info.org/tiki-index.php?page=Asterisk%20AbsoluteTimeout -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P vs SIP3000 x2
On 5/26/05, Andres Paglayan <[EMAIL PROTECTED]> wrote: > > I am about to start building my first ever * production server and would > be nice to have some input from the list. My personal vote would be for the Sipura's. Pro's - It would make failing over to standby box much easier. You could run a small 1u box and not have to worry about PCI requirements. Lightens the load (especially interrupts) on the * box PSTN doesn't have to be located by the * box, just by an ethernet port I think if you poll the archives, you would find problems with both of them. I run a SPA3k and have had no problems with it at all. Just my .02 -Chuji ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo with two IP phones through Asterisk using SIP
I had echo with sjphone, upgraded to x-pro (www.xten.com) today, its $30 but it seems to have eliminated the echo entirely and made my ipaq usable. That may help your problem, although its not free.. sjphone did not seem to have any more options to tweak, asterisk didnt either, so my last choice was to buy software. Sorry if that is the only option for you, but it appears to be that sjphone doesnt have any echo cancelation and on some devices (like my ipaq) the audio in/out is so closely coupled that echo is unavoidable, on any non headset device it also appears to be. On Thu, 2005-05-26 at 18:00 -0400, qrss wrote: > One component of the echo that you are hearing is likely caused by poor > acoustic isolation in the headset of "the caller" that you described. It > might be eliminated by having "the caller" adjust their mic levels down > slightly. Better quality headsets usually have better acoustic isolation > properties too. > > > -Original Message- > From: Davin O'Neill > Sent: Thu, May 26, 2005 8:49 am > > I have Asterisk running on my LAN with softphone clients (SJPhone) and > > Cisco > 7940/60s, all using SIP. I also have a few remote sites connecting to my > Asterisk server. I am getting an echo back of my voice when talking with > one particular site. The caller does not hear an echo on their end. All > calls on the LAN or to other sites do not produce an echo. When the > > caller > places his SJPhone on mute there is no echo. The caller is using a > > standard > PC headset and not a speakerphone. I've tried turning on echocancel and > echotraining in the Zapata configuration, but it had no effect. > > The website below states that two IP phones going through Asterisk should > not result in a noticeable echo, instead this is more common when > > connecting > to PSTN lines. The echo is about 500ms, faint but distracting. Any > suggestions would be greatly appreciated. > > http://www.voip-info.org/wiki-Asterisk+Echo+Avoidance > > Thanks! > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoiPSupply Dot Com
About this business with the distraction of the Cisco licenses... Seems most have put the brunt of the focus of my gripe on that. It's not the licenses, that's like 10% of the problem. One can always buy licenses... But the 16 man hours I wasted waiting for the across-town overnight shipment vastly outweighed the cost of the licenses. The main gripe is: It's the fact that the next-day delivery for across town was two days late. Add to that the fact that it was shipped to the Bill-To address and not the Ship-To address. Thereby causing me to bill out 16 hours of my time (which isn't cheap), for sitting on my hands. THAT IS THE PROBLEM! I thought it was worth mentioning the SIP licenses so others would be in the know as well. Seems Gregory reckons himself an expert on HTML design... About the KTE site: I don't author the ken-ton.com site. Ken-Ton doesn't specialize in HTML authoring, graphics, or the like. What Ken-Ton does specialize in is electronics engineering and research. We've been using VOIP since 1995 combined with Wireless to link the two facilities. Ken-Ton.com has been on the Internet as early as 1992, a subdomain of exucom.com at the time. So, yes... The web page is a might bit outdated, matter of fact I doubt that it's been touched or updated since 1996. But... With that ugly page, it's still been in business since 1975, and is doing quite well I might add. PS: Ken-Ton is currently looking for someone to re-vamp the site, but it's not a real high priority. However, if someone reading this has a web portfolio with some references you might try contacting Joe about it. He's mentioned it to me several times this year, but I have bigger fish to fry than look for some HTML authors. SUMMARY: Voip Supply - Good, however if your project is time critical your mileage may vary. Tech Support is awesome. They don't do 30 day terms, so if you're thinking of quantity purchases you may consider another distributor. I'll still buy stuff from them, and I'll still recommend them for the occasional user who wants to get a Sipura, or Digium card to play with. But if it's time critical I can't recommend them. Does that clear things up any??? At 02:35 AM 5/26/2005, you wrote: Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="_=_NextPart_001_01C561BD.0B0A3F02" OMG! Has anyone tried to visit www.ken-ton.com? It's a laugh! Explains his whole email. Heh... Greg From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Karl J. Vesterling Sent: Wednesday, May 25, 2005 1:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoiPSupply Dot Com We just bought (a couple weeks ago) Qty 6 Cisco Phones from them. There was no licenses with the phones... Go figure... So, evidently they're supposed to be new, and one would expect they come with the license, but alas no license. We discovered this during our initial purchase of a few Sipura Units and (2) Cisco 7960 Phones. When we purchased the (6) additional Cisco 7960's it would seem that no only was there no license (as expected), but the phone itself was configured with an unknown password and they weren't consistent with the Firmware versions installed. Needless to say upgrading these to the latest SIP Firmware was quite the difficult task. Also worthy of note is that I flew to the location where we were deploying these units (Buffalo NY, which oddly enough is the home town of Voip-Supply) and they had shipped the purchased items to the Bill-To address instead of the Ship-To address. Add to that the packages were a day late, and since the Bill-To address is residential they didn't arrive until 4:55PM on a Friday. I had arrived, and anticipated their arrival Thursday AM. In short, I wound up sitting on my hands with my thumb stuck in an extremely uncomfortable place for 16 hours eagerly anticipating the arrival of the purchased items. Here's the kicker. Since we purchased two MediaTrix 1204's I was expecting to be able to get in touch with their technical support if I had difficulties. Alas, Mediatrix business hours are 9:00AM - 5:00PM Mon - Fri Eastern. No dice there... Can you say "Show Stopper" ??? So, not only did I sit on my hands for 16 hours, but this caused me TONS of grief since my regular 9 - 5 job here in DC occupies that spectrum of my time, and the Thursday and Friday when I was expecting delivery was personal time that I took off from work to do this side project. SUMMARY: If you call Voip-Supply and the person that answers the call is named Damon, ask for someone else. On the other hand, feel free to recommend Damons incredibly astounding and unique customer service abilities to all your competitors. At 12:37 AM 5/18/2005, you wrote: I tried calling their toll free number and toll number last week in the morning and afternoon and was handed a recording saying this number is no longer in service. The web sit
Re: [Asterisk-Users] Cisco Call Manager & Asterisk for Voicemail
Well, I have my CM and my Cisco phones leaving and retrieving VM from *. :-) The challenge now is the MWI. Shaun's notes are great! I discovered the version of H323 is not playing well with asterisk-oh323. It looks like I'm going to need to gut the h323 config on my * box and start with a set of known working (interoperable) versions. It looks like if the MWI are working then the * VM call handlers/auto attendants will also work. Has anyone got * doing Auto Attendants for a Call Manager? I'm going to read a few tutorials on H.323 and OH.323. Back soon. Thanks Scott Paul Davidson wrote: Date: Thu, 26 May 2005 16:50:02 +1000 From: Shaun Ewing <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Cisco Call Manager & Asterisk for Voicemail To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1 On 5/26/05, Scott Herrick <[EMAIL PROTECTED]> wrote: BUMP It's CM 3.3.6 MAN that would be sweet if * could take the place of Unity! Anybody? :-) I've got it working with Callmanager 3.3(5) and Asterisk (connected with chan_oh323). Not totally integrated - one still needs to set call forwarding (busy/no answer) on each extension that needs voicemail, but MWI works and so does the messages button (eg: on 7960G) to retrieve VM. If somebody can tell me how to send a call in Callmanager to (for example) extension 27000 when 7000 is unavailable by checking the voicemail box (rather than entering an individual number for each extension), it'll be perfect. I can share my progress so far if it will be beneficial. -Shaun You've done the hard bits. The bad news is that, under CCM, there's really not much in the way of VM configuration. You should set up the VM Pilot stuff to your extension for the Asterisk voicemail- this allows you to click the 'voicemail' box on each extension rather than keying it in- but you still have to touch each extension. You can use their automated tools to make systemwide changes to all extensions- but I don't trust them at all, and I don't think that would help you in this case. I'd love to see how you configured the MWI and how you've set your dialplan- from the way it looks, you're using a different extension for each mailbox. Theoretically, there should be fields on the PDUs from h.323 that show the forwarding number- that's the way Unity does it- and you go into VM for the forwarding number, not for the extension dialed. I'm not sure without playing if any of the h323 channel drivers make the forwarding number available as a channel variable- if they don't, it should be a relatively trivial patch, assuming CCM sends it across (which I'm pretty sure it does- again, time to set some debugs and watch the PDUs). -pbd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do I diagnose the problem in this Asterisk test session with FWD?
=
SJphone Log
Outgoing SIP session
Respondent: (sip:[EMAIL PROTECTED])
Remote client:
Started: May 26 16:33
Accepted: no
Ended: May 26 16:34
End reason: Call rejected: 503 Service Unavailable
===
Asterisk Debug
Executing Dial("SIP/2201-a83e", "IAX2/:@iax2.fwdnet.net/612|60|r")
in new stack
-- Called :@iax2.fwdnet.net/612
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 00014ms SCall: 3 DCall: 0 [65.39.205.121:4569]
VERSION : 2
CALLED NUMBER : 612
CALLING NAME: MyName
LANGUAGE: en
USERNAME:
FORMAT : 4
CAPABILITY : 63494
ADSICPE : 2
DATE TIME : 179995695
Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
Timestamp: 00019ms SCall: 1 DCall: 0 [65.39.205.121:4569]
USERNAME:
REFRESH : 60
Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 00014ms SCall: 3 DCall: 0 [65.39.205.121:4569]
VERSION : 2
CALLED NUMBER : 612
CALLING NAME: MyName
LANGUAGE: en
USERNAME:
FORMAT : 4
CAPABILITY : 63494
ADSICPE : 2
DATE TIME : 179995695
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: LAGRQ
Timestamp: 10019ms SCall: 1 DCall: 0 [65.39.205.121:4569]
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: LAGRQ
Timestamp: 10006ms SCall: 3 DCall: 0 [65.39.205.121:4569]
Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: LAGRQ
Timestamp: 10019ms SCall: 1 DCall: 0 [65.39.205.121:4569]
Tx-Frame Retry[002] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ
Timestamp: 00019ms SCall: 1 DCall: 0 [65.39.205.121:4569]
USERNAME:
REFRESH : 60
=
IAX CONF
register => :[EMAIL PROTECTED]
[iaxfwd]
type=user
auth=rsa
inkeys=freeworlddialup
disallow=all
context=fromiaxfwd
context=fwd-out
EXTENSIONS CONF
FWDUSERID1= ; your calling number
FWDCIDNAME="MyName"; your caller id
FWDPASSWORD=secret ; your password
FWDRINGS=sip/2201 ; the phone to ring
FWDVMBOX=2201 ; the VM box for this user
FWDPREFIX=8
[inbound-fwd]
include => daytime-fwd|9:00-21:00|*|*
include => nighttime-fwd|21:00-9:00|*|*
;
; Outbound
;
[fwd-out]
exten => _8.,1,SetCallerID(${FWDCIDNAME})
exten => _8.,2,Dial(IAX2/${FWDUSERID1}:
[EMAIL PROTECTED]/${EXTEN:1},60,r)
exten => _8.,3,Congestion
exten => _8.,4,Macro(fastbusy)
exten => _8.,5,Hangup
[fromiaxfwd]
exten => ${FWDUSERID1},1,Goto(inbound-fwd,s,1)
exten => ${FWDUSERID1},2,Voicemail2,u${FWDVMBOX}
exten => ${FWDUSERID1},102,Voicemail2,b${FWDVMBOX}
exten => ${FWDUSERID1},3,Hangup
[dialout]
include => fwd-out
include => fromiaxfwd
Thanks
___
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Re: [Asterisk-Users] Echo with two IP phones through Asterisk using SIP
One component of the echo that you are hearing is likely caused by poor acoustic isolation in the headset of "the caller" that you described. It might be eliminated by having "the caller" adjust their mic levels down slightly. Better quality headsets usually have better acoustic isolation properties too. -Original Message- From: Davin O'Neill Sent: Thu, May 26, 2005 8:49 am I have Asterisk running on my LAN with softphone clients (SJPhone) and > Cisco 7940/60s, all using SIP. I also have a few remote sites connecting to my Asterisk server. I am getting an echo back of my voice when talking with one particular site. The caller does not hear an echo on their end. All calls on the LAN or to other sites do not produce an echo. When the > caller places his SJPhone on mute there is no echo. The caller is using a > standard PC headset and not a speakerphone. I've tried turning on echocancel and echotraining in the Zapata configuration, but it had no effect. The website below states that two IP phones going through Asterisk should not result in a noticeable echo, instead this is more common when > connecting to PSTN lines. The echo is about 500ms, faint but distracting. Any suggestions would be greatly appreciated. http://www.voip-info.org/wiki-Asterisk+Echo+Avoidance Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PHP/AGI Problem
On 5/26/05, Benjamin West <[EMAIL PROTECTED]> wrote: > Michael, > The version, in the context of Jon's problem, was irrelevant. Jon's > problem was due to a small bug in his code, and not related to PHPAGI. Yeah. I was just wondering what version people were using since 2.0 hasn't been formally released yet. I spoke to the developer and he suggested 2.0. Thanks, Michael > > Ben > > On 5/26/05, Michael Stearne <[EMAIL PROTECTED]> wrote: > > On 5/26/05, Jon Farmer <[EMAIL PROTECTED]> wrote: > > > > > > Now the script loops forever while the user is connected and exits if > > > the user hangs up. > > > > > > Thanks to everyone who helped me out, much appreciated. > > > > > > > Jon, > > > > What version of PHPAGI are you using? I am starting a PHPAGI app and > > want to know whether to use 1.12 or 2.0CVS. > > > > Thanks, > > Michael > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] YET Another echo issue PRI CARD Any help accepted :-)
Since I have had multiple requests I am posting my supplier here. Hope this is not off topic but they really helped me out and others seem to be interested. Vendor = ditech communications Product I used was their Quad 2 T1, link is below http://www.ditechcom.com/platforms/tdmEchoCancellers.html you may contact the salesman I used at [EMAIL PROTECTED] On May 26, 2005, at 4:13 PM, [EMAIL PROTECTED] wrote: Couple thoughts. first - I hope you have reversed your files listed and zaptel is really zapata and vice versa second - always stop and start * when making changes to zapata - reload does not cut it - (some experience here) third - I wound up with a fairly echo free system using the software echo cancel within * but it was not good enough for a commercial application. I opted out for hardware echo cancelling via DSP and spent the bucks. They have been very good and I was able to turn off on * and reduce its load. My largest intermittant source of echo was traceable to various LEC/ IXC who were clueless on proper installation procedures. I was able to work with some to remedy but having them keep popping up and receiving customer complaints made it worth spending the $$$. good luck On May 26, 2005, at 12:14 PM, Ronald Hartmann wrote: Good Day all, I have a Fractional PRI connected to my Asterisk Box via a T100P card. When I initiate a call out to phone number 123- the call sounds great no echo what so ever. If the person at 123- hangs up and calls me right back (same handset on both sides) same trunk line The call always has echo on it. The Asterisk sip extension hears them selves echoing. The remote party does not notice any difference. I have tried all the following. #define CONFIG_ZAPTEL_MMX Then tried each of the following types of echo cancellations. #define ECHO_CAN_MARK #define ECHO_CAN_MARK2 with and without #define AGGRESSIVE_SUPPRESSOR #define ECHO_CAN_MARK3 I am completely at a loss on how to get rid of this echo problem. The system is completely useless for incoming calls, as it currently stands. Is there a Digium card that handles echo better? Are there any asterisk compatible cards with hardware echo cancellation available? Thanks Ron [Zapata.conf] span=1,1,0,esf,b8zs bchan=19-23 # set this to 1-15,17-31 for E1 dchan=24 # set this to 16 for E1 defaultzone=us loadzone=us [Zaptel.conf] [channels] language=en signalling=pri_cpe switchtype=national pridialplan=unknown echocancel=yes echocancelwhenbridged=yes (tried no) echotraining=400 (tried 800 also) usecallerid=yes callerid=asreceived overlapdial=yes immediate=no group=0,1 context=from-pstn channel => 19-23 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PHP/AGI Problem
Michael, The version, in the context of Jon's problem, was irrelevant. Jon's problem was due to a small bug in his code, and not related to PHPAGI. Ben On 5/26/05, Michael Stearne <[EMAIL PROTECTED]> wrote: > On 5/26/05, Jon Farmer <[EMAIL PROTECTED]> wrote: > > > > Now the script loops forever while the user is connected and exits if > > the user hangs up. > > > > Thanks to everyone who helped me out, much appreciated. > > > > Jon, > > What version of PHPAGI are you using? I am starting a PHPAGI app and > want to know whether to use 1.12 or 2.0CVS. > > Thanks, > Michael > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] new cisco ip video phone?
This looks pretty interesting:
www.ineen.com
Rebranded Eyebeam softphone.
You can call an ineen user with this in your dialplan:
exten => _5.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
I tried to call my own Asterisk server by using [EMAIL PROTECTED] syntax and
I got 401/Unauthorized which is exactly what I expected to get. I added the
ineen account to sip.conf and I got 407/Proxy Authentication Required.
h. Password problem, eh? I'll fire up Ethereal and see what's going on
there.
Haven't tried the video yet. Anyone else tried this thing?
-Original Message-
From: Dean Collins [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 26, 2005 2:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] new cisco ip video phone?
I currently am offering a $3,000 bounty for multiparty video
conferencing on asterisk, there have been 4 people contact me but only 1
of them is actually doing any work on this to my knowledge.
The nearest alternative that I can recommend is smilingtiger, it costs
about 10,000 for up to 10 seats and 36,000 unlimited.
Let me know if you have the budget for it and I'll email you some info
on it.
Cheers,
Dean
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Colin Anderson
> Sent: Thursday, 26 May 2005 3:52 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] new cisco ip video phone?
>
> Too bad. We just shelled out big $$$ for a hosted videoconference ASP
> account and it totally sucks. They even surcharge us for audio, so we
use
> MeetMe instead. At least the Asterisk part works good :- ) >
>
> -Original Message-
> From: Dean Collins [mailto:[EMAIL PROTECTED]
> Sent: Thursday, May 26, 2005 1:19 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion; Lex
Lethol
> Subject: RE: [Asterisk-Users] new cisco ip video phone?
>
>
> I've played with the dlink eyebeam but only for ip to ip calling not
> used with asterisk.
>
> It's crap.
>
> Dean
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
[mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Colin Anderson
> > Sent: Thursday, 26 May 2005 3:12 PM
> > To: 'Lex Lethol'; 'Asterisk Users Mailing List - Non-Commercial
> > Discussion'
> > Subject: RE: [Asterisk-Users] new cisco ip video phone?
> >
> > I noticed several obviously fake screens on '24' and passing
reference
> in
> > the script to "intrusion detection systems" by Cloe, all with the
> Cisco
> > logo
> > featured prominently; this is undoubtedly product placement payola.
I
> > suppose it's possible that Cisco is deploying some preproduction
sets
> to
> > the
> > '24' set to build some hype in the '24' demographic, but occam's
razor
> > suggests that this is just some cisco phone shells with a standard
LCD
> > inside slaved to played back video or a videocamera at the other
end,
> or
> > maybe even inserted digitally post-production.
> >
> > I'd like to hear if anyone on the list has tried the Dlink EyeBeam
> thingy.
> > At least, *that's* a product that is shipping.
> >
> > -Original Message-
> > From: Lex Lethol [mailto:[EMAIL PROTECTED]
> > Sent: Thursday, May 26, 2005 12:40 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] new cisco ip video phone?
> >
> >
> > Hey all,
> >
> > I took some screenshots of the video feed... If you look closely at
> > the upper right part of the 'phone' it says Cisco IP Phone and
barely
> > a 7xxx something. It may be a fake just for showing cisco capable
of
> > doing IP telephony but who knows.. decide for yourself ;)
> >
> > http://lethol.com/blog/ciscoIP.jpg
> > http://lethol.com/blog/ciscoIP2.jpg
> > http://lethol.com/blog/ciscoIP3.jpg
> > http://lethol.com/blog/ciscoIP4.jpg
> > http://lethol.com/blog/ciscoIP5.jpg
> > http://lethol.com/blog/ciscoIP6.jpg
> >
> > Lethol
> >
> >
> >
> > On 5/26/05, Mailing List <[EMAIL PROTECTED]> wrote:
> > > Any chance it's the phone mentioned here?
> > >
> > > http://voxilla.com/voxstory134.html
> > >
> > >
> > > _
> > > Mobilcom
> > > http://www.mobilcom.net
> > >
> > >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED] On Behalf Of Lex
> Lethol
> > > Sent: Thursday, May 26, 2005 2:02 AM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: [Asterisk-Users] new cisco ip video phone?
> > >
> > > Hi,
> > >
> > > Just finished watching the season finale of '24' the TV series.
> > > Throughout the series they have been showcasing Cisco hardware
> > > especially Cisco IP phones (7970's).
> > >
> > > On the last episode or two they showed what seemed to me a new
cisco
> IP
> > > video phone. It stands just as a 12" lcd screen with the cisco
> > > branding/logo and letters just as the 79xx series.
> > >
> > > I wonder if this is a new cisco model thats ready to roll out.
[Asterisk-Users] dhcp vars, mediatrix 1204's
I have been deploying a bunch of sip gateways that I configure via snmp. I have noticed that a lot of the variables I need to set, can be set via dhcp. I like to just put common entries into my dhcpd.conf file, like: option some-variable-name some-variable-value example: option sip-server 192.168.0.1 option sip-port 5060 How do I know what some-variable-name should be in my dhcpd.conf file that will map to some snmp mib variable? I have peeked at the mediatrix mibs and docs and can not seem to find what I am looking for. I am guessing the dhcp client in the gateway is parsing dhcp packets, looking for option names. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PHP/AGI Problem
On 5/26/05, Jon Farmer <[EMAIL PROTECTED]> wrote: > > Now the script loops forever while the user is connected and exits if > the user hangs up. > > Thanks to everyone who helped me out, much appreciated. > Jon, What version of PHPAGI are you using? I am starting a PHPAGI app and want to know whether to use 1.12 or 2.0CVS. Thanks, Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] YET Another echo issue PRI CARD Any help accepted :-)
Couple thoughts. first - I hope you have reversed your files listed and zaptel is really zapata and vice versa second - always stop and start * when making changes to zapata - reload does not cut it - (some experience here) third - I wound up with a fairly echo free system using the software echo cancel within * but it was not good enough for a commercial application. I opted out for hardware echo cancelling via DSP and spent the bucks. They have been very good and I was able to turn off on * and reduce its load. My largest intermittant source of echo was traceable to various LEC/ IXC who were clueless on proper installation procedures. I was able to work with some to remedy but having them keep popping up and receiving customer complaints made it worth spending the $$$. good luck On May 26, 2005, at 12:14 PM, Ronald Hartmann wrote: Good Day all, I have a Fractional PRI connected to my Asterisk Box via a T100P card. When I initiate a call out to phone number 123- the call sounds great no echo what so ever. If the person at 123- hangs up and calls me right back (same handset on both sides) same trunk line The call always has echo on it. The Asterisk sip extension hears them selves echoing. The remote party does not notice any difference. I have tried all the following. #define CONFIG_ZAPTEL_MMX Then tried each of the following types of echo cancellations. #define ECHO_CAN_MARK #define ECHO_CAN_MARK2 with and without #define AGGRESSIVE_SUPPRESSOR #define ECHO_CAN_MARK3 I am completely at a loss on how to get rid of this echo problem. The system is completely useless for incoming calls, as it currently stands. Is there a Digium card that handles echo better? Are there any asterisk compatible cards with hardware echo cancellation available? Thanks Ron [Zapata.conf] span=1,1,0,esf,b8zs bchan=19-23 # set this to 1-15,17-31 for E1 dchan=24 # set this to 16 for E1 defaultzone=us loadzone=us [Zaptel.conf] [channels] language=en signalling=pri_cpe switchtype=national pridialplan=unknown echocancel=yes echocancelwhenbridged=yes (tried no) echotraining=400 (tried 800 also) usecallerid=yes callerid=asreceived overlapdial=yes immediate=no group=0,1 context=from-pstn channel => 19-23 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk x PROLIANT ML 150 G2 SATA
Hi All, I´m tryingo to install asterisk in an PROLIANT ML 150 G2 SATA and can´t make it work because linux cant recognize the Hd (HP 160 mb). No drivers for Centos ...Red Hat... i´t´s drivig me crazy.. Someone have a tip? if i make change it to SCSI i think it will work but not sure about. Thanks Miklos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to setup Dundi in Asterisk?
I have followed the steps, in the wiki, but still no luck which Directory's should they be in. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Wednesday, May 25, 2005 8:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to setup Dundi in Asterisk? Paul Dracevich wrote: >I have set it up, but I get an error, to do with the keys, if I can get >past that part I will have no problems setting up the mappings, dial >rules etc. > >Do u have any ideas, on this error? > > Have you generated the keys? Are they in the right directory? bye Ronald >-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] On Behalf Of Ronald >Wiplinger >Sent: Tuesday, May 24, 2005 8:48 PM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: [Asterisk-Users] How to setup Dundi in Asterisk? > >I subscribed to the dundi mailing list, but so far I have not got a >single message from there. Is there a message archive? > >I want to setup DUNDI. I have a peering agreemrent, but what is next? >I copied from the wiki all parts, but still I am a little bit lost. Has >anybody setup DUNDI? > > >bye > >Ronald > >___ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >___ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for inexpensive phone to use with Asterisk with message light and a button that will let me play new messages
Any phone with a programable voicemail/messages button will do. To name a few: Panasonic TSC14B around $40 is a 2 line phone that has a voicemail button that can be programmed to dial an extension that dials voicemail (and with the s option there is no need for password). Grandstream BT (I hate this phone) And any other VOIP phone that has a programmable message button. On 5/26/05, Kellner, Peter <[EMAIL PROTECTED]> wrote: > I'm wanting to have a phone at home next to the garage door that when my > bride comes home, she can see that there is a new message, push a button > and have the messages played to her. Otherwise, she will not let me > install asterisk on my home line. > > Can someone suggest relatively inexpensive hardware that will do this > for me (us)? > > Thanks, > > -Peter > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] tds_CDR and MS SQL Server troubleshooting
In the SQL server MMC, you can expand SQL server > Management > Process Info and see if your defined user is listed in there after you try to hit SQL server. If not, it is most certainly a rights problem. Quick and dirty way to determine if it's a rights problem is to use the sql server SA account as the login. If it works, then it's a rights problem. hth -Original Message- From: PA [mailto:[EMAIL PROTECTED] Sent: Thursday, May 26, 2005 2:16 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] tds_CDR and MS SQL Server troubleshooting I am trying to get * to write CDR records to an MS SQL table. I am used the freeTDS set up (not using UNIX ODBC), and it appears to be making the connection to the database, but I make a call, I get this error: WARNING[3369]: cdr_tds.c:180 tds_log: Reconnected to SQL database. ERROR[3369]: cdr_tds.c:191 tds_log: Failed to insert Call Data Records into SQL database. I've assigned all rights to the cdr table in SQL to the user. I am not sure how to troubleshoot this from this point. Does anyone know else these errors are? Thanks for any direction. Phil ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: tds_CDR and MS SQL Server troubleshooting
In article <[EMAIL PROTECTED]>, PA <[EMAIL PROTECTED]> wrote: > I am trying to get * to write CDR records to an MS SQL table. I am used the > freeTDS set up > (not using UNIX ODBC), and it appears to be making the connection to the > database, but I > make a call, I get this error: > > WARNING[3369]: cdr_tds.c:180 tds_log: Reconnected to SQL database. > ERROR[3369]: cdr_tds.c:191 tds_log: Failed to insert Call Data Records into > SQL database. > > I've assigned all rights to the cdr table in SQL to the user. I am not sure > how to > troubleshoot this from this point. Does anyone know else these errors are? > Thanks for any > direction. It's probably a database or SQL Server configuration problem. The module works, because I'm using it in production. If you are using safe_asterisk to start up, put the following lines into /usr/sbin/safe_asterisk near the top, just after the DUMPDROP definition: TDSDUMP=/tmp/tdsdump.$$.log TDSDUMPCONFIG=/tmp/tdsdumpconfig.$$.log export TDSDUMP TDSDUMPCONFIG Then look for those files in the /tmp directory when asterisk is running. They will contain a lot more debugging information from the FreeTDS libraries. If you're not using safe_asterisk, then just call the above lines from your shell before you invoke asterisk. Hope this helps Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk con X-lite : Register Ok but no calls (404 Not found)
Hi all,
I'm working on an implementation of VoIP en Linux.
I have a Debian Suse (*.*.*.173) with an * and a X-lite client and a
Red Hat 9.0 (*.*.*.172) with another softphone X-lite.
Both of the softphones are registering and appear in the peers (sip
show peers) with the good parameters of address and port.
If I try to make a call, * receive the INVITE request and send a 404
NOT FOUND answer.
I can't understand why asterisk doesn't found the users if they are registred...
It's making a "Scheduling Call Destruction".
My config files are :
sip.conf :
[general]
>>context=default; Default context for incoming calls
>>recordhistory=yes; Record SIP history by default
>>port=5060; UDP Port to bind to (SIP standard port is 5060)
>>bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)
>>srvlookup=yes
>>
>>[]
>>;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
>>;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
>>type=friend
>>username=
>>secret=
>>callerid="Thibaud" <>
>>host=dynamic
>>context=from-sip
>>allow=ulaw
>>qualify=yes
>>
>>[]
>>;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
>>;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
>>type=friend
>>username=
>>secret=
>>callerid="Florentin" <>
>>host=dynamic
>>context=from-sip
>>allow=ulaw
>>qualify=yes
extensions.conf :
>>[bogon-calls
>>exten => _.,1,Congestion
>>
>>[from-sip]
>>
>>exten => ,1,Dial(SIP/,20)
>>exten => ,2,Voicemail(u)
>>exten => ,102,Voicemail(b)
>>exten => ,103,Hangup
>>
>>exten => ,1,Dial(SIP/,20)
>>exten => ,2,Voicemail(u)
>>exten => ,102,Voicemail(b)
>>exten => ,103,Hagup
>>
>>exten => ,1,VoicemailMain(${CALLERIDNUM})
The critical SIP exchange is :
SEND TIME: 440651449
SEND >> *.*.*.173:5060
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
*.*.*.172:5060;rport;branch=z9hG4bK5E6AA1A5168F2672CCC77E03310CA049
From: Asterisk ;tag=93980267
To:
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 30470 INVITE
Proxy-Authorization: Digest
username="",realm="asterisk",nonce="2c887956",response="79eb7583cec4b45e867189dfa7d515dd",uri="sip:[EMAIL
PROTECTED]"
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105d
Content-Length: 285
v=0
o= 440651420 440651437 IN IP4 *.*.*.172
s=X-Lite
c=IN IP4 *.*.*.172
t=0 0
m=audio 1 RTP/AVP 0 8 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
RECEIVE TIME: 440651467
RECEIVE << *.*.*.173:5060
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP *.*.*.172:5060;branch=z9hG4bK5E6AA1A5168F2672CCC77E03310CA049
From: Asterisk ;tag=93980267
To: ;tag=as6c9ced81
Call-ID: [EMAIL PROTECTED]
CSeq: 30470 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
--
Romain Barrallon
- Etudiant en Télécommunications, Services et Usages à l'INSA de Lyon (France)
- Estudiante de intercambio en la Universidad Tecnica Federico Santa
Maria de Valparaíso (Chile)
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[Asterisk-Users] TDM400P vs SIP3000 x2
Dear List, Besides price, ~$300 against ~$200, Is there any pros and/or cons on using one or the other approach to provide 2 FXSs and 2 FXOs (plus 4 IP phone extensions)? I am about to start building my first ever * production server and would be nice to have some input from the list. Many thanks, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] tds_CDR and MS SQL Server troubleshooting
I am trying to get * to write CDR records to an MS SQL table. I am used the freeTDS set up (not using UNIX ODBC), and it appears to be making the connection to the database, but I make a call, I get this error: WARNING[3369]: cdr_tds.c:180 tds_log: Reconnected to SQL database. ERROR[3369]: cdr_tds.c:191 tds_log: Failed to insert Call Data Records into SQL database. I've assigned all rights to the cdr table in SQL to the user. I am not sure how to troubleshoot this from this point. Does anyone know else these errors are? Thanks for any direction. Phil ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Limiting maximum runtime of echo test
Hello everybody, is it possible to limit the maximum runtime of the command "echo"? Regards Bastian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] new cisco ip video phone?
I currently am offering a $3,000 bounty for multiparty video conferencing on asterisk, there have been 4 people contact me but only 1 of them is actually doing any work on this to my knowledge. The nearest alternative that I can recommend is smilingtiger, it costs about 10,000 for up to 10 seats and 36,000 unlimited. Let me know if you have the budget for it and I'll email you some info on it. Cheers, Dean > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Colin Anderson > Sent: Thursday, 26 May 2005 3:52 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] new cisco ip video phone? > > Too bad. We just shelled out big $$$ for a hosted videoconference ASP > account and it totally sucks. They even surcharge us for audio, so we use > MeetMe instead. At least the Asterisk part works good :- ) > > > -Original Message- > From: Dean Collins [mailto:[EMAIL PROTECTED] > Sent: Thursday, May 26, 2005 1:19 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion; Lex Lethol > Subject: RE: [Asterisk-Users] new cisco ip video phone? > > > I've played with the dlink eyebeam but only for ip to ip calling not > used with asterisk. > > It's crap. > > Dean > > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Colin Anderson > > Sent: Thursday, 26 May 2005 3:12 PM > > To: 'Lex Lethol'; 'Asterisk Users Mailing List - Non-Commercial > > Discussion' > > Subject: RE: [Asterisk-Users] new cisco ip video phone? > > > > I noticed several obviously fake screens on '24' and passing reference > in > > the script to "intrusion detection systems" by Cloe, all with the > Cisco > > logo > > featured prominently; this is undoubtedly product placement payola. I > > suppose it's possible that Cisco is deploying some preproduction sets > to > > the > > '24' set to build some hype in the '24' demographic, but occam's razor > > suggests that this is just some cisco phone shells with a standard LCD > > inside slaved to played back video or a videocamera at the other end, > or > > maybe even inserted digitally post-production. > > > > I'd like to hear if anyone on the list has tried the Dlink EyeBeam > thingy. > > At least, *that's* a product that is shipping. > > > > -Original Message- > > From: Lex Lethol [mailto:[EMAIL PROTECTED] > > Sent: Thursday, May 26, 2005 12:40 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] new cisco ip video phone? > > > > > > Hey all, > > > > I took some screenshots of the video feed... If you look closely at > > the upper right part of the 'phone' it says Cisco IP Phone and barely > > a 7xxx something. It may be a fake just for showing cisco capable of > > doing IP telephony but who knows.. decide for yourself ;) > > > > http://lethol.com/blog/ciscoIP.jpg > > http://lethol.com/blog/ciscoIP2.jpg > > http://lethol.com/blog/ciscoIP3.jpg > > http://lethol.com/blog/ciscoIP4.jpg > > http://lethol.com/blog/ciscoIP5.jpg > > http://lethol.com/blog/ciscoIP6.jpg > > > > Lethol > > > > > > > > On 5/26/05, Mailing List <[EMAIL PROTECTED]> wrote: > > > Any chance it's the phone mentioned here? > > > > > > http://voxilla.com/voxstory134.html > > > > > > > > > _ > > > Mobilcom > > > http://www.mobilcom.net > > > > > > > > > -Original Message- > > > From: [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] On Behalf Of Lex > Lethol > > > Sent: Thursday, May 26, 2005 2:02 AM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: [Asterisk-Users] new cisco ip video phone? > > > > > > Hi, > > > > > > Just finished watching the season finale of '24' the TV series. > > > Throughout the series they have been showcasing Cisco hardware > > > especially Cisco IP phones (7970's). > > > > > > On the last episode or two they showed what seemed to me a new cisco > IP > > > video phone. It stands just as a 12" lcd screen with the cisco > > > branding/logo and letters just as the 79xx series. > > > > > > I wonder if this is a new cisco model thats ready to roll out. It > looks > > > great, but then again, I doubt they will support SIP on it (at least > on > > > release) > > > > > > Anyone else know anything on this? > > > > > > Lethol > > > > > > ___ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > __
Re: [Asterisk-Users] CRM integration (was RE: CallerID)
And another update. This time I made a .tar.gz file. The file is called cidXML.tar.gz and is on http://michiel.vanbaak.info/Files/ The file contains the agi and some php scripts to do the notification. I rewrote it so it is not Dutch anymore. I also added comments so you know what's going on and why. If you have questions, ask. This weekend I'm spending with family, so I might be slow in responding to email. Greetings, -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence." ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk connecting to Nortel 1000 with SIP
Did you search for Nortel at www.voip-info.org? Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Thursday, May 26, 2005 12:52 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk connecting to Nortel 1000 with SIP Was wondering if anyone has asterisk connected to a Nortel 1000 using SIP? Did some searching on google and did not find anything. Is it straight forward, quirks, does not work? Thanks, Jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] new cisco ip video phone?
Too bad. We just shelled out big $$$ for a hosted videoconference ASP account and it totally sucks. They even surcharge us for audio, so we use MeetMe instead. At least the Asterisk part works good :- ) > -Original Message- From: Dean Collins [mailto:[EMAIL PROTECTED] Sent: Thursday, May 26, 2005 1:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Lex Lethol Subject: RE: [Asterisk-Users] new cisco ip video phone? I've played with the dlink eyebeam but only for ip to ip calling not used with asterisk. It's crap. Dean > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Colin Anderson > Sent: Thursday, 26 May 2005 3:12 PM > To: 'Lex Lethol'; 'Asterisk Users Mailing List - Non-Commercial > Discussion' > Subject: RE: [Asterisk-Users] new cisco ip video phone? > > I noticed several obviously fake screens on '24' and passing reference in > the script to "intrusion detection systems" by Cloe, all with the Cisco > logo > featured prominently; this is undoubtedly product placement payola. I > suppose it's possible that Cisco is deploying some preproduction sets to > the > '24' set to build some hype in the '24' demographic, but occam's razor > suggests that this is just some cisco phone shells with a standard LCD > inside slaved to played back video or a videocamera at the other end, or > maybe even inserted digitally post-production. > > I'd like to hear if anyone on the list has tried the Dlink EyeBeam thingy. > At least, *that's* a product that is shipping. > > -Original Message- > From: Lex Lethol [mailto:[EMAIL PROTECTED] > Sent: Thursday, May 26, 2005 12:40 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] new cisco ip video phone? > > > Hey all, > > I took some screenshots of the video feed... If you look closely at > the upper right part of the 'phone' it says Cisco IP Phone and barely > a 7xxx something. It may be a fake just for showing cisco capable of > doing IP telephony but who knows.. decide for yourself ;) > > http://lethol.com/blog/ciscoIP.jpg > http://lethol.com/blog/ciscoIP2.jpg > http://lethol.com/blog/ciscoIP3.jpg > http://lethol.com/blog/ciscoIP4.jpg > http://lethol.com/blog/ciscoIP5.jpg > http://lethol.com/blog/ciscoIP6.jpg > > Lethol > > > > On 5/26/05, Mailing List <[EMAIL PROTECTED]> wrote: > > Any chance it's the phone mentioned here? > > > > http://voxilla.com/voxstory134.html > > > > > > _ > > Mobilcom > > http://www.mobilcom.net > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Lex Lethol > > Sent: Thursday, May 26, 2005 2:02 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [Asterisk-Users] new cisco ip video phone? > > > > Hi, > > > > Just finished watching the season finale of '24' the TV series. > > Throughout the series they have been showcasing Cisco hardware > > especially Cisco IP phones (7970's). > > > > On the last episode or two they showed what seemed to me a new cisco IP > > video phone. It stands just as a 12" lcd screen with the cisco > > branding/logo and letters just as the 79xx series. > > > > I wonder if this is a new cisco model thats ready to roll out. It looks > > great, but then again, I doubt they will support SIP on it (at least on > > release) > > > > Anyone else know anything on this? > > > > Lethol > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk connecting to Nortel 1000 with SIP
Was wondering if anyone has asterisk connected to a Nortel 1000 using SIP? Did some searching on google and did not find anything. Is it straight forward, quirks, does not work? Thanks, Jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking for inexpensive phone to use with Asterisk with message light and a button that will let me play new messages
And to point out additional info, the backlight for the entire phone flashes when the mailbox it is programmed to monitor has a message. MUCH easier to see than a little flashing red light. Robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Thursday, May 26, 2005 3:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Looking for inexpensive phone to use with Asterisk with message light and a button that will let me play new messages Why does it need to be near the garage? Isn't she house trained? And what's wrong with the grandstream bt101? You can program the message key with the access number and code to access the voicemail? > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Kellner, Peter > Sent: Thursday, 26 May 2005 3:04 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Looking for inexpensive phone to use with > Asteriskwith message light and a button that will let me play new messages > > I'm wanting to have a phone at home next to the garage door that when my > bride comes home, she can see that there is a new message, push a button > and have the messages played to her. Otherwise, she will not let me > install asterisk on my home line. > > Can someone suggest relatively inexpensive hardware that will do this > for me (us)? > > Thanks, > > -Peter > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip extension logon failed problem
Yes, when I configure the clients of Asterisk I always use LAN (192.168.2.1) as proxy server. I also use sentences like register => xxx:[EMAIL PROTECTED]/1234 to register to other Sip Servers. When I set bindaddr=192.168.2.1 all the Accounts registering to other Sip Servers will logon failed even though the WAN is good. I really cannot figure out how to make Asterisk logon to its own Sip Servers and make its clients logon to itself also unless bindaddr is not 0.0.0.0 Cound you help me on this problem, please? Thank you so much. Be Well. lanfei --- Gentian Bajraktari <[EMAIL PROTECTED]> wrote: > Well you must only use the LAN IP as the proxy ip > when configuring the > clients. Then use the LANIP to be the bindaddress at > sip.conf. This will > make the SIP proxy available to the intermal clients > even if the WAN > interface goes down. > > Gentian > - Original Message - > From: "lanfei chen" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial > Discussion" > > Sent: Thursday, May 26, 2005 5:55 PM > Subject: Re: [Asterisk-Users] sip extension logon > failed problem > > > > several IPs, > > 67.105.168.51, > > 67.105.168.52, > > 61.235.77.130, > > 218.104.232.147 > > > > Thank you for your response. > > > > lanfei > > > > --- Gentian Bajraktari <[EMAIL PROTECTED]> > > wrote: > >> Which IP are you using on the client > configurations > >> for proxy server? > >> > >> Gentian > >> > >> - Original Message - > >> From: "lanfei chen" <[EMAIL PROTECTED]> > >> To: "Asterisk Users Mailing List - Non-Commercial > >> Discussion" > >> > >> Sent: Thursday, May 26, 2005 4:08 PM > >> Subject: Re: [Asterisk-Users] sip extension logon > >> failed problem > >> > >> > >> > Hi Gentian, > >> > Thank you so much. > >> > But it doesn't work. > >> > My problem is in the case of bindaddr=0.0.0.0 > >> > If I set bindaddr=LAN-IP, it cannot logon to > its > >> sip > >> > proxy server. > >> > If I set bindaddr=WAN-IP, the sip extensions > >> cannot > >> > logon. > >> > > >> > Did you test and succeed? Wish you can give me > >> more > >> > information. > >> > > >> > Thank you. > >> > Be well. > >> > > >> > lanfei > >> > > >> > > >> > --- Gentian Bajraktari > <[EMAIL PROTECTED]> > >> > wrote: > >> >> Check if the SIP.conf is configured to bind > into > >> the > >> >> ip address of LAN or > >> >> not? You have to define the internal address > >> there, > >> >> not the WAN ip address. > >> >> > >> >> RG, > >> >> > >> >> Gentian > >> >> > >> >> - Original Message - > >> >> From: "lanfei chen" <[EMAIL PROTECTED]> > >> >> To: "Asterisk Users Mailing List - > Non-Commercial > >> >> Discussion" > >> >> > >> >> Sent: Wednesday, May 25, 2005 2:52 PM > >> >> Subject: [Asterisk-Users] sip extension logon > >> failed > >> >> problem > >> >> > >> >> > >> >> > Hi All, > >> >> >Now I configured a linux box as a router. > >> And I > >> >> > installed Asterisk on it. > >> >> >My problem is whenever the WAN is offline > >> all > >> >> the > >> >> > sip extensions will logon failed. My sip > >> >> extensions > >> >> > are connected to Asterisk through LAN. Why > the > >> LAN > >> >> > side sip phones cannot logon when WAN is > >> offline. > >> >> > > >> >> > Who knows the reason and solution? > >> >> > Thank you so much. > >> >> > I really appreciate. > >> >> > > >> >> > Be Well. > >> >> > > >> >> > lanfei > >> >> > > >> >> > > >> >> > > >> >> > __ > >> >> > Do you Yahoo!? > >> >> > Yahoo! Small Business - Try our new > Resources > >> site > >> >> > http://smallbusiness.yahoo.com/resources/ > >> >> > > ___ > >> >> > Asterisk-Users mailing list > >> >> > Asterisk-Users@lists.digium.com > >> >> > > >> >> > >> > > >> > > > http://lists.digium.com/mailman/listinfo/asterisk-users > >> >> > To UNSUBSCRIBE or update options visit: > >> >> > > >> >> > >> > > >> > > > http://lists.digium.com/mailman/listinfo/asterisk-users > >> >> > > >> >> > > >> >> > >> >> > >> >> > ___ > >> >> Asterisk-Users mailing list > >> >> Asterisk-Users@lists.digium.com > >> >> > >> > > >> > > > http://lists.digium.com/mailman/listinfo/asterisk-users > >> >> To UNSUBSCRIBE or update options visit: > >> >> > >> >> > >> > > >> > > > http://lists.digium.com/mailman/listinfo/asterisk-users > >> >> > >> > > >> > > >> > > >> > __ > >> > Do you Yahoo!? > >> > Yahoo! Small Business - Try our new Resources > site > >> > http://smallbusiness.yahoo.com/resources/ > >> > ___ > >> > Asterisk-Users mailing list > >> > Asterisk-Users@lists.digium.com > >> > > >> > > > http://lists.digium.com/mailman/listinfo/asterisk-users > >> > To UNSUBSCRIBE or update options visit: > >> > > >> > > > http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > >> > > >> > >> > >> ___ > >> Asterisk-Us
Re: [Asterisk-Users] new cisco ip video phone?
Anyone tried the Packet8 Videophones ? I would guess that leadtek is providing the "non-branded" version now ? [], Dean Collins wrote: I've played with the dlink eyebeam but only for ip to ip calling not used with asterisk. It's crap. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Thursday, 26 May 2005 3:12 PM To: 'Lex Lethol'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] new cisco ip video phone? I noticed several obviously fake screens on '24' and passing reference in the script to "intrusion detection systems" by Cloe, all with the Cisco logo featured prominently; this is undoubtedly product placement payola. I suppose it's possible that Cisco is deploying some preproduction sets to the '24' set to build some hype in the '24' demographic, but occam's razor suggests that this is just some cisco phone shells with a standard LCD inside slaved to played back video or a videocamera at the other end, or maybe even inserted digitally post-production. I'd like to hear if anyone on the list has tried the Dlink EyeBeam thingy. At least, *that's* a product that is shipping. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] YET Another echo issue PRI CARD Any help acc epted :-)
I have found that the audio is hot from some carriers and low on others. I have found that this is causing the echocanclers problems. Before I reduce it down by 3db I will see if some of the problem in in the Supura . Andrew Kohlsmith wrote: On May 26, 2005 01:58 pm, Colin Anderson wrote: I have had good success fiddling with the txgain and rxgain values in zapata.conf on my PRI. In my setup, cranking the gain down a LOT eliminated most of the echo, and training the users to turn down the gain on their handsets did the rest. It's true, with a PRI, that gains are cranked across the board. Turning the gain down solves a lot of echo problems, with negligible effect on voice quality. There should be *NO* reason to adjust tx/rxgain on a PRI or ANY digital connection! The fact that adjusting it down 10% worked suggests that the telco switch is boosting the signal for some unknown reason. Use the technique outlined in this message: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html (It works VERY well and is very straightforward, thank you Kris, I reference this all the time!) Again if you're screwing with gains on PRI you have bigger problems, I think. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote Voicemail Notifier / enter Dialplan onSIPRegister
I've added the patch for CVS HEAD 05-02-2005 to the Mantis entry ( http://bugs.digium.com/view.php?id=4371) last night as well as the patch to CVS HEAD as of 5/24 -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking for inexpensive phone to use with Asterisk with message light and a button that will let me play new messages
Why does it need to be near the garage? Isn't she house trained? And what's wrong with the grandstream bt101? You can program the message key with the access number and code to access the voicemail? > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Kellner, Peter > Sent: Thursday, 26 May 2005 3:04 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Looking for inexpensive phone to use with > Asteriskwith message light and a button that will let me play new messages > > I'm wanting to have a phone at home next to the garage door that when my > bride comes home, she can see that there is a new message, push a button > and have the messages played to her. Otherwise, she will not let me > install asterisk on my home line. > > Can someone suggest relatively inexpensive hardware that will do this > for me (us)? > > Thanks, > > -Peter > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] new cisco ip video phone?
I've played with the dlink eyebeam but only for ip to ip calling not used with asterisk. It's crap. Dean > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Colin Anderson > Sent: Thursday, 26 May 2005 3:12 PM > To: 'Lex Lethol'; 'Asterisk Users Mailing List - Non-Commercial > Discussion' > Subject: RE: [Asterisk-Users] new cisco ip video phone? > > I noticed several obviously fake screens on '24' and passing reference in > the script to "intrusion detection systems" by Cloe, all with the Cisco > logo > featured prominently; this is undoubtedly product placement payola. I > suppose it's possible that Cisco is deploying some preproduction sets to > the > '24' set to build some hype in the '24' demographic, but occam's razor > suggests that this is just some cisco phone shells with a standard LCD > inside slaved to played back video or a videocamera at the other end, or > maybe even inserted digitally post-production. > > I'd like to hear if anyone on the list has tried the Dlink EyeBeam thingy. > At least, *that's* a product that is shipping. > > -Original Message- > From: Lex Lethol [mailto:[EMAIL PROTECTED] > Sent: Thursday, May 26, 2005 12:40 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] new cisco ip video phone? > > > Hey all, > > I took some screenshots of the video feed... If you look closely at > the upper right part of the 'phone' it says Cisco IP Phone and barely > a 7xxx something. It may be a fake just for showing cisco capable of > doing IP telephony but who knows.. decide for yourself ;) > > http://lethol.com/blog/ciscoIP.jpg > http://lethol.com/blog/ciscoIP2.jpg > http://lethol.com/blog/ciscoIP3.jpg > http://lethol.com/blog/ciscoIP4.jpg > http://lethol.com/blog/ciscoIP5.jpg > http://lethol.com/blog/ciscoIP6.jpg > > Lethol > > > > On 5/26/05, Mailing List <[EMAIL PROTECTED]> wrote: > > Any chance it's the phone mentioned here? > > > > http://voxilla.com/voxstory134.html > > > > > > _ > > Mobilcom > > http://www.mobilcom.net > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Lex Lethol > > Sent: Thursday, May 26, 2005 2:02 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [Asterisk-Users] new cisco ip video phone? > > > > Hi, > > > > Just finished watching the season finale of '24' the TV series. > > Throughout the series they have been showcasing Cisco hardware > > especially Cisco IP phones (7970's). > > > > On the last episode or two they showed what seemed to me a new cisco IP > > video phone. It stands just as a 12" lcd screen with the cisco > > branding/logo and letters just as the 79xx series. > > > > I wonder if this is a new cisco model thats ready to roll out. It looks > > great, but then again, I doubt they will support SIP on it (at least on > > release) > > > > Anyone else know anything on this? > > > > Lethol > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CRM integration (was RE: CallerID)
On 18:48, Wed 25 May 05, Anton Krall wrote: > Michiel. > > To help better understand what xmlhttp does with asterisk, could you modify > your tgx file and strip everything related to db and make a sort of hello > world script? In fact, something small like connecting to the manager (of > course, we each have to put our config there) and maybe issue a sip show > peers and show how this can be done and update in realtime via web? > > What do you think? This will elp grasp the concept of xmlhttp and connecting > to the manager. > Anton, My script is not connecting to the manager interface. The php script is run as agi script as first when a call comes in. The php script will know the callerID of the calling party, that is something the AGI interface provides to the script. Then the AGI script connects to my CRM database to do a lookup on the phone number. If it finds a match it will put the companyName and the records ID in a temporary table in the same database. Of course instead of storing it in a database it can also write a little xml file. I think I will do that in the next release of our CRM app (Where this code is taken from). I altered the agi script to simplify it a little and I added comments to explain what is being done. I will also do this to the php script that generates the callerID notification in my CRM app. I will upload the new agi to my server as soon as this mail is sent. Just for the record: http://michiel.vanbaak.info/Files/ Have fun, > > > |-Original Message- > |From: [EMAIL PROTECTED] > |[mailto:[EMAIL PROTECTED] On Behalf Of > |Michiel van Baak > |Sent: Mi?rcoles, 25 de Mayo de 2005 02:04 p.m. > |To: asterisk-users@lists.digium.com > |Subject: Re: [Asterisk-Users] CRM integration (was RE: CallerID) > | > |On 10:17, Wed 25 May 05, Rusty Shackleford wrote: > |> > -Original Message- > |> > From: [EMAIL PROTECTED] > |> > [mailto:[EMAIL PROTECTED] On Behalf > |Of Anton > |> > Krall > |> > Sent: Wednesday, May 25, 2005 7:41 AM > |> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > |> > Subject: RE: [Asterisk-Users] CRM integration (was RE: CallerID) > |> > > |> > > |> > It doesn?t seem to be complicated but for example, the things that > |> > bother me are refreshes, I don?t want to use meta > |refreshes for this > |> > monitoring webpage every X seconds, rather, use something more > |> > realtime... Any ideas? > |> > |> And that's the real trick. Web browsers, unless they are > |instructed to > |> do otherwise, don't DO anything once they've completed > |loading a page. > |> So without instructing them to refresh, they aren't going to > |be aware > |> of a server-side change, such as an incoming call. For that, you're > |> going > | > |This is not true. If it was for pure HTML only, yes, you are > |correct. But with javascript you can start a timer and execute > |a javascript function every once in a while. > |If this javascript loads an XML document off the server, > |you're there ;) > | > | > |> to have to have some way of sending a message TO the client machine, > |> have it received by that machine, and have that client machine take > |> the desired action (pop up an incoming call dialog, load a contact > |> record, etc.). > |> > | > |Have a look at the tgz file I posted earlier > | > |> -- > |> No virus found in this outgoing message. > |> Checked by AVG Anti-Virus. > |> Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: > |> 05/20/2005 > |> > |> > |> ___ > |> Asterisk-Users mailing list > |> Asterisk-Users@lists.digium.com > |> http://lists.digium.com/mailman/listinfo/asterisk-users > |> To UNSUBSCRIBE or update options visit: > |>http://lists.digium.com/mailman/listinfo/asterisk-users > | > |-- > |Michiel van Baak > |http://lunteren.vanbaak.info > |[EMAIL PROTECTED] > |GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D > | > |"Two of the most famous products of Berkeley are LSD and BSD. > |I don't think that this is a coincidence." > | > |___ > |Asterisk-Users mailing list > |Asterisk-Users@lists.digium.com > |http://lists.digium.com/mailman/listinfo/asterisk-users > |To UNSUBSCRIBE or update options visit: > | http://lists.digium.com/mailman/listinfo/asterisk-users > | > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence." ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.
RE: [Asterisk-Users] Asterisk@home - mysql login
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Quintin Sent: Thursday, May 26, 2005 1:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] [EMAIL PROTECTED] - mysql login Hi all,whats the root password for [EMAIL PROTECTED] db, to login from the consel? Thx Q That would be a question that should be directed toward the [EMAIL PROTECTED] forum…. http://sourceforge.net/forum/?group_id=123387 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on 64 bit Linux
Title: Asterisk on 64 bit Linux Hello All, After reading a recent exchange off the list archive, I began to wonder if there is any more information on running Asterisk on 64 bit Linux distros. Of question… 1. What performance increase if any? 2. Stability issues? 3. Any issues with the Digium or Sangoma hardware drivers? 4. And finally, is this documented anywhere? So far, all I have found is… http://lists.digium.com/pipermail/asterisk-users/2005-May/107058.html RPMs I have found here… http://rpm.pbone.net/index.php3/stat/4/idpl/1867424/com/asterisk-1.0.7-10.el3.at.x86_64.rpm.html Does anyone know where more info on Asterisk on 64 bit Linux can be found? Specifically the user experience and performance, stability, etc… Thanks, Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ICD Usage Examples
Does anyone use ICD in a production machine? Any problems with it handling a moderate load with multiple queues? Is there a comparsion list of ACD and ICD anywhere? --johann ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM > * rings once and goes fast busy
Title: Message Hi all, Any help with this would be appreciated: we are running 0.7.1 OH323 for asterisk...here is the scenario... TDM > H323 Gateway > Asterisk > SIP Endpoint the SIP phone rings one time and then the origination TDM phone goes fast busy now...on the other hand...this works: SIP Enpoint > Asterisk > H323 Gateway > TDM > works PERFECT...practically 1 second PDD...splendid sound quality anyone with any ideas i would appreciate it... jeromy Global reach, local touch... Jeromy GrimmettCEO Comuniquémonos, Inc. / SA1212 South Hampton DriveAlexandria, LA 71301 [EMAIL PROTECTED]IM: MSN: [EMAIL PROTECTED]http://www.comuniquemonos.com tel: fax: mobile: +593 (4) 287 3854(501) 646-0680+593 (9) 366 6521 Add me to your address book... Want a signature like this? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] new cisco ip video phone?
I noticed several obviously fake screens on '24' and passing reference in the script to "intrusion detection systems" by Cloe, all with the Cisco logo featured prominently; this is undoubtedly product placement payola. I suppose it's possible that Cisco is deploying some preproduction sets to the '24' set to build some hype in the '24' demographic, but occam's razor suggests that this is just some cisco phone shells with a standard LCD inside slaved to played back video or a videocamera at the other end, or maybe even inserted digitally post-production. I'd like to hear if anyone on the list has tried the Dlink EyeBeam thingy. At least, *that's* a product that is shipping. -Original Message- From: Lex Lethol [mailto:[EMAIL PROTECTED] Sent: Thursday, May 26, 2005 12:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] new cisco ip video phone? Hey all, I took some screenshots of the video feed... If you look closely at the upper right part of the 'phone' it says Cisco IP Phone and barely a 7xxx something. It may be a fake just for showing cisco capable of doing IP telephony but who knows.. decide for yourself ;) http://lethol.com/blog/ciscoIP.jpg http://lethol.com/blog/ciscoIP2.jpg http://lethol.com/blog/ciscoIP3.jpg http://lethol.com/blog/ciscoIP4.jpg http://lethol.com/blog/ciscoIP5.jpg http://lethol.com/blog/ciscoIP6.jpg Lethol On 5/26/05, Mailing List <[EMAIL PROTECTED]> wrote: > Any chance it's the phone mentioned here? > > http://voxilla.com/voxstory134.html > > > _ > Mobilcom > http://www.mobilcom.net > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Lex Lethol > Sent: Thursday, May 26, 2005 2:02 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] new cisco ip video phone? > > Hi, > > Just finished watching the season finale of '24' the TV series. > Throughout the series they have been showcasing Cisco hardware > especially Cisco IP phones (7970's). > > On the last episode or two they showed what seemed to me a new cisco IP > video phone. It stands just as a 12" lcd screen with the cisco > branding/logo and letters just as the 79xx series. > > I wonder if this is a new cisco model thats ready to roll out. It looks > great, but then again, I doubt they will support SIP on it (at least on > release) > > Anyone else know anything on this? > > Lethol > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for inexpensive phone to use with Asterisk with message light and a button that will let me play new messages
I'm wanting to have a phone at home next to the garage door that when my bride comes home, she can see that there is a new message, push a button and have the messages played to her. Otherwise, she will not let me install asterisk on my home line. Can someone suggest relatively inexpensive hardware that will do this for me (us)? Thanks, -Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting a couple DS0's to a wildcard
There's a project to use asterisk to replace an old analog pbx for a hospital in Mexico. The person in charge wants to do a trial before making a large investment. For this test, he is willing to buy a Wildcard TE410P. Hes got a proposal from his pstn provider to convert his analog lines to digital ones, but I have some doubts. Let's say he converts only 2 or three of the lines. He would get two or three DS0's, wouldn't he? Now, if that were the case, would it be possible to connect these to the TE410P? Is it just that only three time slots are used and the others are blank, or signalling is different between the cases you don't have a full E1? Thanks in advance Jaime Atahualpa Lopez Sollano ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dropping frame of G.729 since we already have a VAD frame at the end
I have this showing on my cli while being in a call. Then connection gets broken. Can someone tell me what it means ? Dropping frame of G.729 since we already have a VAD frame at the end Thank you in advance. Bartosz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] new cisco ip video phone?
Hey all, I took some screenshots of the video feed... If you look closely at the upper right part of the 'phone' it says Cisco IP Phone and barely a 7xxx something. It may be a fake just for showing cisco capable of doing IP telephony but who knows.. decide for yourself ;) http://lethol.com/blog/ciscoIP.jpg http://lethol.com/blog/ciscoIP2.jpg http://lethol.com/blog/ciscoIP3.jpg http://lethol.com/blog/ciscoIP4.jpg http://lethol.com/blog/ciscoIP5.jpg http://lethol.com/blog/ciscoIP6.jpg Lethol On 5/26/05, Mailing List <[EMAIL PROTECTED]> wrote: > Any chance it's the phone mentioned here? > > http://voxilla.com/voxstory134.html > > > _ > Mobilcom > http://www.mobilcom.net > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Lex Lethol > Sent: Thursday, May 26, 2005 2:02 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] new cisco ip video phone? > > Hi, > > Just finished watching the season finale of '24' the TV series. > Throughout the series they have been showcasing Cisco hardware > especially Cisco IP phones (7970's). > > On the last episode or two they showed what seemed to me a new cisco IP > video phone. It stands just as a 12" lcd screen with the cisco > branding/logo and letters just as the 79xx series. > > I wonder if this is a new cisco model thats ready to roll out. It looks > great, but then again, I doubt they will support SIP on it (at least on > release) > > Anyone else know anything on this? > > Lethol > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Soft Video phone for Asterisk usage
Try Eyebeam from xten.com. I usethe phone part but haven't tried the video yet. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Ronald Wiplinger wrote: I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] YET Another echo issue PRI CARD Any help acc epted :-)
On May 26, 2005 01:58 pm, Colin Anderson wrote: > I have had good success fiddling with the txgain and rxgain values in > zapata.conf on my PRI. In my setup, cranking the gain down a LOT eliminated > most of the echo, and training the users to turn down the gain on their > handsets did the rest. It's true, with a PRI, that gains are cranked across > the board. Turning the gain down solves a lot of echo problems, with > negligible effect on voice quality. There should be *NO* reason to adjust tx/rxgain on a PRI or ANY digital connection! The fact that adjusting it down 10% worked suggests that the telco switch is boosting the signal for some unknown reason. Use the technique outlined in this message: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html (It works VERY well and is very straightforward, thank you Kris, I reference this all the time!) Again if you're screwing with gains on PRI you have bigger problems, I think. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Using zap channels on 2 different servers
You use pattern matching in extensions.conf to determine where the call is
to go. In North America, you dial 1+3+3+4 digits for LD and 3+4 for local.
In Europe, you dial 3+3+3 between countries or 3+3 in the same country* ,
so,
EUROPEAN SERVER:
[outbound]
exten => _9XXX,1,Dial(IAX2/europeuser:[EMAIL PROTECTED]/${exten})
'Local north america calling 3+4 digit
exten =>
_9XXX,1,Dial(IAX2/europeuser:[EMAIL PROTECTED]/${exten}) 'LD
north america calling 1+3+3+4 digit)
AMERICA SERVER:
[outbound]
exten => _9XX,1,Dial(IAX2/americauser:[EMAIL PROTECTED]/${exten})
'Local europe calling 3+3 digit
exten =>
_9X,1,Dial(IAX2/americauser:[EMAIL PROTECTED]/${exten})
'Cross-country europe calling 3+3+3 digit)
In this example, the X's mean any digit. If you count the number of X's, you
get 7 digits for NA local, 11 for LD, 6 for European local, 9 for European
LD.
In both of these examples, dialing 9 before the number is assumed. Also, it
is assumed that IAX.CONF on both servers is set up correctly to dump the
inbound IAX call into a context that has access to the ZAP channels of the
respective server.
You could also modify this so if they dial 8 first, for example, it would
always go to Europe or vice versa. But then it becomes a user training
issue. If you just tell them to dial 9 + the number then Asterisk will
figure out the rest.
hth
*It's my understanding that there are exceptions in certain European
countries. You will have to figure out those execptions.
-Original Message-
From: Martin Roy [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 26, 2005 11:49 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Using zap channels on 2 different servers
Let say I have a server located in Europe and one in North America.
The 2 servers are connected together with iax2.
Both server are connected to phone lines in there own country.
If I want that when a user call a north american phone number from
the server in Europe it use a zap channel on the server located in
North America and also if someone in North America dial an European
phone number it use a zap channel on the server in Europe how can I
setup this?
Currently it's configure that when a user press 9 and enter the
number to dial it takes a local zap channels to dial out.
I figured easily how to call a remote extension as I use a different
range of extensions for both site but can I use remote zap channels
the same way?
Thanks
Martin
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Re: [Asterisk-Users] SER Help
> Do you not have access to the Asterisk Machine's NAT > box? I do __ Do you Yahoo!? Yahoo! Small Business - Try our new Resources site http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_dial_rev5
Im trying to install the app_dial_rev5.diff patch from http://bugs.digium.com/bug_view_page.php?bug_id=0002905 so I can do call screening. Im not a programmer so Im unsure whats going on. Any help would be appreciated.
[EMAIL PROTECTED] asterisk-1.0.7]# patch -p0
patching file pbx.c
Hunk #1 FAILED at 4935.
Hunk #2 succeeded at 5357 with fuzz 2 (offset -57 lines).
1 out of 2 hunks FAILED -- saving rejects to file pbx.c.rej
patching file include/asterisk/pbx.h
Hunk #1 FAILED at 578.
1 out of 1 hunk FAILED -- saving rejects to file include/asterisk/pbx.h.rej
patching file apps/app_dial.c
Hunk #1 succeeded at 67 (offset -1 lines).
Hunk #2 succeeded at 461 (offset -29 lines).
Hunk #3 FAILED at 469.
Hunk #4 succeeded at 660 (offset -8 lines).
Hunk #5 succeeded at 958 (offset -50 lines).
Hunk #6 succeeded at 1009 (offset -8 lines).
Hunk #7 succeeded at 981 (offset -50 lines).
1 out of 7 hunks FAILED -- saving rejects to file apps/app_dial.c.rej
app_dial.rej -
***
*** 463,471
char *varname;
int vartype;
char *outbound_group = NULL;
-
int digit = 0;
time_t start_time, answer_time, end_time;
if (!data) {
ast_log(LOG_WARNING, "Dial requires an argument (technology1/number1&technology2/number2...|optional timeout|options)\n");
--- 469,478
char *varname;
int vartype;
char *outbound_group = NULL;
+ char *macro_result = NULL, *macro_transfer_dest = NULL;
int digit = 0;
time_t start_time, answer_time, end_time;
+ struct ast_app *app = NULL;
if (!data) {
ast_log(LOG_WARNING, "Dial requires an argument (technology1/number1&technology2/number2...|optional timeout|options)\n");
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RE: [Asterisk-Users] multiples broadvoice lines
On Thu, 2005-05-26 at 12:48 -0500, Jay Milk wrote: > Nothing wrong with putting them all in the same context and using Goto > -- in fact, I've been using that with nine SIP lines from three > different providers and a dozen incoming DIDs from two IAX providers. > Why, you ask? Because you have your ALL call-distribution nicely > contained in a single file -- extensions.conf. I never said there was anything wrong with that if that is what you choose to do, however I did say that if you do not choose to put them all in the same context and have them all go to different contexts instead asterisk ignores your feeble request and does what it wants. And that in my book qualifies as a bug. If I set a unique context for each account, the mere fact they are all from the same sip proxy should not override that. It does not if they are from different proxies so it makes no sense that it does when they are the same proxy. I think it was either a lazy programmer or a bad sort algorithm (perhaps an if that doesnt have enough compares for unique connection information?) Granted this is a rare occurance for testing purposes, if a test case was not created to test for this problem specifically it would not be uncovered until someone used asterisk to try to do exactly this. I just feel that people should have choice, simple little freedoms to do their extensions.conf however they want, and not be forced to put them all in the same context if they do not want to. Maybe my feelings on freedom and choice are too far out there and the better solution is to do it one way because that way is best for one person. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip extension logon failed problem
Well you must only use the LAN IP as the proxy ip when configuring the clients. Then use the LANIP to be the bindaddress at sip.conf. This will make the SIP proxy available to the intermal clients even if the WAN interface goes down. Gentian - Original Message - From: "lanfei chen" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, May 26, 2005 5:55 PM Subject: Re: [Asterisk-Users] sip extension logon failed problem several IPs, 67.105.168.51, 67.105.168.52, 61.235.77.130, 218.104.232.147 Thank you for your response. lanfei --- Gentian Bajraktari <[EMAIL PROTECTED]> wrote: Which IP are you using on the client configurations for proxy server? Gentian - Original Message - From: "lanfei chen" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, May 26, 2005 4:08 PM Subject: Re: [Asterisk-Users] sip extension logon failed problem > Hi Gentian, > Thank you so much. > But it doesn't work. > My problem is in the case of bindaddr=0.0.0.0 > If I set bindaddr=LAN-IP, it cannot logon to its sip > proxy server. > If I set bindaddr=WAN-IP, the sip extensions cannot > logon. > > Did you test and succeed? Wish you can give me more > information. > > Thank you. > Be well. > > lanfei > > > --- Gentian Bajraktari <[EMAIL PROTECTED]> > wrote: >> Check if the SIP.conf is configured to bind into the >> ip address of LAN or >> not? You have to define the internal address there, >> not the WAN ip address. >> >> RG, >> >> Gentian >> >> - Original Message - >> From: "lanfei chen" <[EMAIL PROTECTED]> >> To: "Asterisk Users Mailing List - Non-Commercial >> Discussion" >> >> Sent: Wednesday, May 25, 2005 2:52 PM >> Subject: [Asterisk-Users] sip extension logon failed >> problem >> >> >> > Hi All, >> >Now I configured a linux box as a router. And I >> > installed Asterisk on it. >> >My problem is whenever the WAN is offline all >> the >> > sip extensions will logon failed. My sip >> extensions >> > are connected to Asterisk through LAN. Why the LAN >> > side sip phones cannot logon when WAN is offline. >> > >> > Who knows the reason and solution? >> > Thank you so much. >> > I really appreciate. >> > >> > Be Well. >> > >> > lanfei >> > >> > >> > >> > __ >> > Do you Yahoo!? >> > Yahoo! Small Business - Try our new Resources site >> > http://smallbusiness.yahoo.com/resources/ >> > ___ >> > Asterisk-Users mailing list >> > Asterisk-Users@lists.digium.com >> > >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> > To UNSUBSCRIBE or update options visit: >> > >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> > >> >> >> ___ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > __ > Do you Yahoo!? > Yahoo! Small Business - Try our new Resources site > http://smallbusiness.yahoo.com/resources/ > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI setup
On Thu, May 26, 2005 at 10:11:53AM -0500, Jay Milk wrote: > You don't know how to compile asterisk? So why'd you ask about PRI? If > you're starting from scratch, google for "Getting Started with Asterisk" > and click on the first link. I've compiled asterisk from CVS head on a Debian Sid system. It works fine with my FXS and FXO interfaces for calls in both directions. I'm ready to move on to the PRI interfaces and found the documentation trail harder to follow. I outlined a document that I'd like to see about PRI installation, configuration, and test. The first steps are kind of obvious to the experienced, nevertheless, they shouldn't be omitted. If I was writing this document, I would create links to other documents that cover these steps. This document: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/x460.html has you building zaptel and asterisk but not libpri. In an Asterisk/PRI HOWTO it would be important to mention building and installing libpri in the correct order. It would also be good to describe how to verify that it was done correctly so one doesn't waste time with an improperly configured system. Thanks for your help so far. -- Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] static database config gui
Hi, Nice work! Is it easy to make it run without mod_perl? Is the source online? best regards, Magnus On 5/26/05, snacktime <[EMAIL PROTECTED]> wrote: > I threw together a web gui for the static database configuration over > the last couple of days. > > I built it using mod perl and the template toolkit. If enough people > show an interest in this I'll put up a distribution, although it could > take a few days. > > The interface is as generic as possible so you can throw pretty much > any asterisk .conf file in and it works. The interface assumes you > already know how to edit the config files. The database schema is the > same as on the wiki. > > I'm working on making it a multi user interface. So that you can have > multiple end users with their own copies of the config files all on > the same server. The separation will be done through a naming > convention that will be applied appropriately. A kind of asterisk > virtual hosting. > > I have a demo setup at the following url: > > http://catalog1.paymentonline.com/voip/demo/index.html > > > One note on the gui. The numbers on the very left are the order of > the statements in the config file. For extensions, when you change > the location of an extension priority the system will automatically > renumber the order and the dialplan automatically. To insert a new > priority in the middle of an extension, use a number with a fraction. > When you add, delete, or update the system will automatically renumber > everything. > > For example if you have the following extension: > > exten => 999,1,Answer > exten => 999,2,Dial > exten => 999,3,Hangup > > And you want to insert a new priority after 1, add the new priority as > 1.5 which when added would give you something like this: > > exten => 999,1,Answer > exten => 999,2,Ringing > exten => 999,3,Dial > exten => 999,4,Hangup > > > Chris > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@home - mysql login
Quintin wrote: Hi all,whats the root password for [EMAIL PROTECTED] db, to login from the consel? Thx Q ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users AFAIK passw0rd Regards Dean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
