Re: SV: [Asterisk-Users] Re: IPswitch: How to use speed dialing?
On Apr 25, 2005, at 10:55 PM, Ronald Wiplinger wrote: Hi Ronald, What happens in your Asterisk box when you press the Speed Dial number in IPS? Can we make it so that you FIRST answer below questions, please? | | Let's try it together: Ronald: wow. Take a breath before you torch a generous developer. IPS works like a charm for me in every way. Seriously, /rg ___ Asterisk-Users mailing list Asterisk-Users@lists.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] astrecipes v2.0
One clarification: On Mon, Apr 25, 2005 at 05:07:31PM +0200, lenz wrote: See http://www.oinko.net/astrecipes All content is licenced as creative commons, so if you got a recipe to spere, feel free to post it. Creative-Commons is a group of licenses. You seem to refer to CreativeCommons Atribution+Share Alike (basically the same idea as GPL, if I read this correctly, and IANAL) -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk integration with Alcatel 4400
Hi Tim, which hardware did you use in the asterisk box for the job? Francesco Tim Connolly [EMAIL PROTECTED] Sent by: To asterisk-users-bo 'Asterisk Users Mailing List - [EMAIL PROTECTED] Non-Commercial Discussion' m.com asterisk-users@lists.digium.com cc 26/04/2005 05.33 Subject RE: [Asterisk-Users] Asterisk integration with Alcatel 4400 Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com I was able to fully integrate my Lucent/Avaya Definity G3. Basically it has a TIE line PRI between the PBX and the *. I had to do some legacy pbx tricks on the Definity to make it send the calls across, but it seems to work pretty well. I would assume the Alcatel could do the same. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, April 25, 2005 4:33 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk integration with Alcatel 4400 I have just finished to look at the interesting documentation linked from voip-info.org entitled: PBX Interoperability: Alcatel 4400 R3.2 PBX with Cisco CallManager with Analog FXS and FXO interfaces as an MGCP Gateway here it is explained how to use FXS and FXO interfaces to do integration. Using * with TDM400P instead of a Cisco 3640 should be possible without big problems, but... ...this is applicable in those cases where there are a small mumber of lines to be routed between the Alcatel PBX and Asterisk, and this will be the solution we will try to set up for a pilot project. If the requirement is to route an entire E1 trunk through *, having in any case the E1 line coming out of the PBX, is it possible to use some Digium hw (ie TE410P) to do the job? In other words, which is the best way to integrate an * PBX into an existing legacy environment? I know it depends on the PBX, its features and its ability to be (further) upgraded both in hw and sw... A short term goal would be to have a number of branch offices (equipped with IP phones) connected to a central Asterisk box through an IP WAN (ADSL or MPLS), and have the asterisk box connected to the central PBX so that any call directed to the root number of the company can be routed by the PBX to * and then to the remote IP phones. Then * could conquer also the typical domain of the legacy PBX, and then ...anyone know what else... Francesco Pellegrini ++ | Frame Srl | | Via Antonio Cantore 62/10 | | 16149 Genova | | Tel. +39 010 8680570| | Fax. +39 010 6591413 | | Cell. +348 2237798 | ++ ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No busy tone when dialing out over ISDN with Polycom 500 IP
Isn't the zapata.conf only for Digium hardware? I use an Eicon Card. Eric Wieling aka ManxPower wrote: Kib Eki wrote: Hi, what do i have to configure to get a busy tone when dialing out over ISDN channel with my Polycom 500 IP? Try priindication = inband in /etc/asterisk/zapata.con ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Distinctive ring on BT100
Hello, Is it possible to make BT100 phones ring in different ways based on where the call is coming from? The general idea is that I need the BT100 ring in 2 different ways depending on whether the call come from Zap1 or Zap2. It's because this system is for a receptionist answering two different phone lines for two separate companies and she needs to know how to greet the person on the other side ... one way that could be useful for her to recognize which line is ringing is by having a different ring tone for each. If BT100 cannot do it .. which phone can? Or is there some alternative way of helping the receptionist in this situation distinguish between the two lines? (Flash Operator Panel would not work well since she would not have it on all the time) Thanks, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.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] astrecipes v2.0
In data Tue, 26 Apr 2005 09:13:52 +0300, Tzafrir Cohen [EMAIL PROTECTED] ha scritto: One clarification: On Mon, Apr 25, 2005 at 05:07:31PM +0200, lenz wrote: See http://www.oinko.net/astrecipes All content is licenced as creative commons, so if you got a recipe to spere, feel free to post it. Creative-Commons is a group of licenses. You seem to refer to CreativeCommons Atribution+Share Alike (basically the same idea as GPL, if I read this correctly, and IANAL) Yes, that's it - see http://www.oinko.net/astrecipes/index.php?n=53 Anyway, all Creative Common licences (AFAIK) share the idea that your content does not become part of the intellectual property of some Evil Corporation or something, that is my main concern when I contribuite to some initiative on the Internet. l. -- Assum est, versa et manduca. ___ Asterisk-Users mailing list Asterisk-Users@lists.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: SV: [Asterisk-Users] Re: IPswitch: How to use speed dialing?
Robert Goodyear wrote:
On Apr 25, 2005, at 10:55 PM, Ronald Wiplinger wrote:
Hi Ronald,
What happens in your Asterisk box when you press the Speed Dial
number in
IPS?
Can we make it so that you FIRST answer below questions, please?
| | Let's try it together:
Ronald: wow. Take a breath before you torch a generous developer. IPS
works like a charm for me in every way.
Seriously,
/rg
For me it doesn't!!!
And IF it works so good for you, why you are not willing / able to fill
the simple three fields out for me, telling me what you think would work.
I ask this question now several times, and get only answers, like it
works, it works for me too
FOR ME IT DOES NOT WORK, ...
So how did yo make it that it works for you???
Questions below to be kindly filled out:
| | Let's try it together:
| 1. Open IPswitch
| 2. Open Extensions tab on top
| 3. Switch to the tab Speed Dials on the bottom
| 4. Fill in:
| Name: [EMAIL PROTECTED]
| Caller Id: Peter
| Visible on Panel: (ticket)
| Exentension Group: Speed Dial Numbers
CLI answers:
| | | Congratualtions, you have successfully installed the Asterisk Open
| Source .
| tgj wrote:
|
|
|
| Hi Ronald,
|
| I must admit I am getting confused now.
|
| I understand that you have a problem getting Speed Dial Buttons to
| work.
| The problem as I understand it is that the calls are placed in the
| wrong
| context.
|
| To solve that problem I have asked you to make sure that you have
typed
| a
| valid context on the configuration page. Have you tried that?
|
| I think thats all you need to do, how do I post an example of that?
| It's a
| fairly easy thing to do.
|
| Thorben
|
|
|
|
|
| What is the right syntax to do that?
| Context for dialing a trunk line is trunkint
| Peter has the phone number 011-234-5678
| How to set it up as a speed dial number? Below are all info you may
| need:
|
| The phone 601 (= Monitor extension) is a Sip phone,
|
| [general]
| context=default; Default context for incoming calls
|
| [601]
| type=friend
| username=601
| secret=dont+tell+you
| canreinvite=no
| host=dynamic
| dtmfmode=rfc2833
| [EMAIL PROTECTED]
| nat=yes
| callgroup=1
| pickupgroup=1
| callerid=Ronald Hotline,601
| qualify=1000
|
|
| extensions.conf
| [default]
| ...
| include = trunkint
| ...
|
| [trunkint]
| ;
| ; International long distance through trunk
| ; . other lines deleted
| exten = _9011Z.,107,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
| exten = _9011Z.,108,hangup
|
|
|
|
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[Asterisk-Users] NO ringback tone for VOIP call to another SIP server
All,
I found that there is no ringback to the caller (a-party) for
VoIP call but when I make call to registered user, I can hear the
ringback tone.
Beloware the debug log for the two cases:
I wonder if anyone who can tell me why?
Thanks.
Raymond
Case 1: no ringback to the caller (a-party) for outbond
VoIP call to another SIP server
Apr 26 07:04:09
VERBOSE[2607]: -- Executing Dial("SIP/30511694-abfa",
"SIP/[EMAIL PROTECTED]") in new stackApr 26 07:04:09 DEBUG[2607]:
Outgoing Call for 99740185293137656Apr 26 07:04:09 DEBUG[2607]:
99740185293137656 is not a local userApr 26 07:04:09
VERBOSE[2607]: -- Called [EMAIL PROTECTED]Apr 26 07:04:09 DEBUG[2607]: (Provisional) Stopping retransmission (but
retaining packet) on '[EMAIL PROTECTED]' Request 102: FoundApr 26 07:04:13 DEBUG[2607]:
(Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: FoundApr 26 07:04:13
VERBOSE[2607]: -- SIP/192.168.11.194-8dc7 is making
progress passing it to SIP/30511694-abfaApr 26 07:04:13 DEBUG[2607]: Ooh,
format changed from unknown to ulawApr 26 07:04:13 DEBUG[2607]: Auto
destroying call '[EMAIL PROTECTED]'Apr 26 07:04:13 DEBUG[2607]: RTP NAT: Using address
192.168.19.241:64868Apr 26 07:04:13 DEBUG[2607]: Oooh, format changed to
8Apr 26 07:04:13 DEBUG[2607]: Ooh, format changed from unknown to
ulawApr 26 07:04:13 DEBUG[2607]: Ooh, format changed from ulaw to
alawApr 26 07:04:15 NOTICE[2607]: RFC3389 support incomplete. Turn off
on client if possibleApr 26 07:04:32 DEBUG[2607]:
update_user_counter(99740185293137656) - decrement outUse counterApr 26
07:04:32 DEBUG[2607]: 99740185293137656 is not a local userApr 26 07:04:32
DEBUG[2607]: Exiting with DIALSTATUS=CANCEL.Apr 26 07:04:32
VERBOSE[2607]: == Spawn extension (siptest02, 85293137656, 1) exited
non-zero on 'SIP/30511694-abfa'Apr 26 07:04:32
VERBOSE[2607]: -- Executing Hangup("SIP/30511694-abfa",
"") in new stackApr 26 07:04:32 VERBOSE[2607]: == Spawn
extension (siptest02, h, 1) exited non-zero on 'SIP/30511694-abfa'Apr 26
07:04:32 DEBUG[2607]: cdr_mysql: inserting a CDR record.Apr 26 07:04:32
DEBUG[2607]: cdr_mysql: SQL command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
VALUES ('2005-04-26 07:04:09','\"cisco 7960\"
30511694','30511694','85293137656','siptest02',
'SIP/30511694-abfa','SIP/192.168.11.194-8dc7','Hangup','',23,0,'NO
ANSWER',3,'')Apr 26 07:04:32 DEBUG[2607]: update_user_counter(30511694) -
decrement inUse counterApr 26 07:04:32 DEBUG[2607]: Acked pending invite
102Apr 26 07:04:32 DEBUG[2607]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: FoundApr 26 07:04:32 DEBUG[2607]:
Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: FoundApr 26 07:04:32 DEBUG[2607]:
99740185293137656 is not a local userApr 26 07:04:32 DEBUG[2607]: Stopping
retransmission on '[EMAIL PROTECTED]' of Response 102: Found
Case 2: When I make call to registered user, I
can hear the ringback tone:
Apr 26 07:05:49 DEBUG[2607]: Auto
destroying call '[EMAIL PROTECTED]'Apr 26 07:05:50 DEBUG[2607]: Setting NAT on RTP to
4Apr 26 07:05:50 DEBUG[2607]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 101: FoundApr 26 07:05:50
DEBUG[2607]: Setting NAT on RTP to 4Apr 26 07:05:50 DEBUG[2607]: Check for
res for 30511694Apr 26 07:05:50 DEBUG[2607]: Call from user '30511694' is 1
out of 0Apr 26 07:05:50 DEBUG[2607]: build_route: Contact hop:
sip:[EMAIL PROTECTED]:5060Apr 26 07:05:50
VERBOSE[2607]: -- Executing Dial("SIP/30511694-581e",
"SIP/30511690|20|tr") in new stackApr 26 07:05:50 DEBUG[2607]: SIMPLE DIAL
(NO URL)Apr 26 07:05:50 DEBUG[2607]: Setting NAT on RTP to 4Apr 26
07:05:50 DEBUG[2607]: Outgoing Call for 30511690Apr 26 07:05:50 DEBUG[2607]:
Call from user '30511690' is 1 out of 0Apr 26 07:05:50
VERBOSE[2607]: -- Called 30511690Apr 26 07:05:50
DEBUG[2607]: (Provisional) Stopping retransmission (but retaining packet) on
'[EMAIL PROTECTED]' Request 102: FoundApr 26 07:05:50 DEBUG[2607]:
(Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: FoundApr 26 07:05:50
VERBOSE[2607]: -- SIP/30511690-adb1 is ringingApr 26
07:06:00 DEBUG[2607]: update_user_counter(30511690) - decrement outUse
counterApr 26 07:06:00 DEBUG[2607]: Exiting with DIALSTATUS=CANCEL.Apr
26 07:06:00 VERBOSE[2607]: == Spawn extension (siptest02, 1690, 1)
exited non-zero on 'SIP/30511694-581e'Apr 26 07:06:00
VERBOSE[2607]: -- Executing Hangup("SIP/30511694-581e",
"") in new stackApr 26 07:06:00 VERBOSE[2607]: == Spawn
extension (siptest02, h, 1) exited non-zero on 'SIP/30511694-581e'Apr 26
07:06:00 DEBUG[2607]: cdr_mysql: inserting a CDR record.Apr 26 07:06:00
DEBUG[2607]: cdr_mysql: SQL command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
VALUES
[Asterisk-Users] VOIP Gateways Asterisk
Hi all, I was just wondering if someone could help me with info on VOIP Gateways. We are planning to do an * install in an apartment building, this building is going to require somewhere in the vacinity of 20 E1 lines (each with 30 voice channels). Short of buying 20 Servers with Digium cards, what are my options in having the E1 lines terminate on some other hardware and then having the calls passed through to Asterisk to perform the PBX type functionality ? I have heard that using some form of VOIP gateway should help, but I really have no idea how this works. Any help would be appreciated. Cheers, Callum ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with long delay. VPN ?
We have a problem whereby once a call has started there is no noticeable delay however as the call continues you notice a delay building up, after say 10mins it becomes really noticeable. The setup is XTEN with VPN (Cisco) - Cisco 7206 - VPN (Cisco) XTEN The RTP and signalling are going across the VPN. Could it be the VPN (encrypt / decrypt) causing the problem ? Has anyone else had this issue and or knows a solution. Thanks as always. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] IAX help
Kanuri, Seshu (Company IT) wrote: If you look at your iax.conf lines as under, you will notice that the two contexts are illegal as they both have same name: I don't believe that part of your advice is correct. I have a number of such entries in my iax.conf and they seem to work without problem. It is actually recommended by some providers that it be done that way. I'll stand corrected on this, but for sure I am using separate peer/user entries without problem. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.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: NO ringback tone for VOIP call to another SIP server
Hi all, To my surprise, I change the Dial statement in extensions.conf from: exten = _852.,1,Dial,SIP/[EMAIL PROTECTED],r to: exten = _852.,1,Dial(SIP/[EMAIL PROTECTED],20,r) I can hear ringback tone now. I don't know why but it just works. Cheers. Raymond - Original Message - From: raymond To: asterisk-users@lists.digium.com Sent: Tuesday, April 26, 2005 3:22 PM Subject: NO ringback tone for VOIP call to another SIP server All, I found that there is no ringback to the caller (a-party) for VoIP call but when I make call to registered user, I can hear the ringback tone. Beloware the debug log for the two cases: I wonder if anyone who can tell me why? Thanks. Raymond ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Error on the Mysql, realtime database HELP soclose so far; .
Hi, Look into your "*.conf" files Res_mysql.conf is a good start Check for user-id password Also check the dbsock=. (the default value did not correspond to my 'default' installation of sql). I have now dbsock= /var/lib/mysql/mysql.sock Look for THAT file in your system. Also dbhost=127.0.0.1 Also check mysql to see if THAT user may connect from THAT machine 127.0.0.1 Good luck I assumed you use mysql and connect with mysql socket on localhost Adapt if you use odbc or another host Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail [EMAIL PROTECTED] image001.jpg___ Asterisk-Users mailing list Asterisk-Users@lists.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] VOIP Gateways Asterisk
Wait until the new card of Digium go out, withit you only will need 1 server more powerful. regards. - Original Message - From: Callum McGillivray [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, April 26, 2005 9:29 AM Subject: [Asterisk-Users] VOIP Gateways Asterisk Hi all, I was just wondering if someone could help me with info on VOIP Gateways. We are planning to do an * install in an apartment building, this building is going to require somewhere in the vacinity of 20 E1 lines (each with 30 voice channels). Short of buying 20 Servers with Digium cards, what are my options in having the E1 lines terminate on some other hardware and then having the calls passed through to Asterisk to perform the PBX type functionality ? I have heard that using some form of VOIP gateway should help, but I really have no idea how this works. Any help would be appreciated. Cheers, Callum ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Re: NO ringback tone for VOIP call to another SIP server
On Tue, 26 Apr 2005, raymond wrote: To my surprise, I change the Dial statement in extensions.conf from: exten = _852.,1,Dial,SIP/[EMAIL PROTECTED],r to: exten = _852.,1,Dial(SIP/[EMAIL PROTECTED],20,r) I can hear ringback tone now. I don't know why but it just works. In the first line you passed th r in the argument reserved for the timeout value. Th options field in Dial is the third argument, not the second. So, you had a timeout of r seconds (invalid) and no ringback option. 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] VOIP Gateways Asterisk
My problem is that this installation is most likely to occur prior to the release of the new card (and definitely prior to it's vigorous testing in the field). If anyone can give me ideas at this point it would be appreciated. Callum Dpto. Tcnico (Softec). wrote: Wait until the new card of Digium go out, withit you only will need 1 server more powerful. regards. - Original Message - From: "Callum McGillivray" [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, April 26, 2005 9:29 AM Subject: [Asterisk-Users] VOIP Gateways Asterisk Hi all, I was just wondering if someone could help me with info on VOIP Gateways. We are planning to do an * install in an apartment building, this building is going to require somewhere in the vacinity of 20 E1 lines (each with 30 voice channels). Short of buying 20 Servers with Digium cards, what are my options in having the E1 lines terminate on some other hardware and then having the calls passed through to Asterisk to perform the PBX type functionality ? I have heard that using some form of VOIP gateway should help, but I really have no idea how this works. Any help would be appreciated. Cheers, Callum ___ Asterisk-Users mailing list Asterisk-Users@lists.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] help to configure sip server asterisk
hi everybody I'm a new Asterisker. I have a very simple configuration : 1 Sip proxy and 2 grandstream 102 in ethernet with private adress sip proxy : 192.168.2.194 ip phone address : 192.168.2.144 192.168.2.195 I want to make a communication between 2 ip phone with the SIP proxy but i have 2 different problems : 1 - a grandstream phone (192.168.2.144 ) can't register with this error : 192.168.2.144 --192.168.2.194 register 192.168.2.194--192.168.2.144 100 trying 192.168.2.194--192.168.2.144 401 unauthorized Registration for '[EMAIL PROTECTED]' timed out, trying again 2 - I can ring the phone with the sip proxy but phone can't make a phone between us and phone can't call the proxy. invite 484 address incomplete someone can help me to find the problem, please I join my config : ---sip.conf [general] ;--- general setup port = 5060 bindaddr = 192.168.2.194 tos = none ;--- codecs setup disallow = all allow = ulaw ;autorise PCMU allow = alaw ;autorise PCMA allow = ilbc ;autorise ILBC ;--- other options ;NETWORK ;localnet = 192.168.2.0 fromdomain = 192.168.2.1 ;---CONTEXT context = from-sip-external ;context = from-sip-internal ;context = default maxexpirey = 3600 srvlookup = yes nat = no ;promiscredir = no ;useragent = Asterisk PBX defaultexpirey = 120 ;trustrpid = no ;musicclass = default [grandstream1] type = friend username = grandstream1 accountcode = grandstream1 dtmfmode = info host =dynamic defaultip = 192.168.2.144 port = 5061 secret = monpassword context = from-sip-internal canreinvite = yes nat = no reinvite = no qualify = yes ;rtnoupdate=no [grandstream2] type = friend username = grandstream2 accountcode = grandstream2 ;callerid=101 dtmfmode = info secret = password host = dynamic defaultip = 192.168.2.195 context = from-sip-internal port = 5060 auth=md5 canreinvite = no nat = no reinvite = no qualify = yes extensions.conf- [general] static=yes writeprotect=yes ; [local] exten = 100,1,Dial(SIP/[EMAIL PROTECTED],10) exten = 101,1,Dial(SIP/[EMAIL PROTECTED],10) ;exten = 2,1,Dial(SIP/2/@192.168.2.194,10) ;- [from-sip-internal] include = local ; rudy73 _ MSN Hotmail : antivirus et antispam gratuits http://www.imagine-msn.com/hotmail/default.aspx?locale=fr-FR ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP/NetMeeting
Does anyone know if it is possible to resolve an IP from outside a small LAN. I would like to be able to specify a SIP client that is outside my office LAN. The problem is that the isp will not provide a static IP that's affordable. I use a DYNDNS.org address with it. When I want to use NetMeeting for desktop sharing, I just ping the DYNDNS address and it gives me the current IP of the remote machine. Is it possible to specify the host name, say billscomputer.dyndns.org for the address of the SIP client in the appropriate .conf file for Asterisk? Thanks, Bill William C. Lohr Jr. Lohr Technologies, LLC www.lohrtechnologies.com(301) 334-8758[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] Distinctive ring on BT100
You can only set 1 distinctive ring if by caller id. There is a tool on the website to record custom ring tone. - Original Message - From: Tomas Florian [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, April 26, 2005 1:39 AM Subject: [Asterisk-Users] Distinctive ring on BT100 Hello, Is it possible to make BT100 phones ring in different ways based on where the call is coming from? The general idea is that I need the BT100 ring in 2 different ways depending on whether the call come from Zap1 or Zap2. It's because this system is for a receptionist answering two different phone lines for two separate companies and she needs to know how to greet the person on the other side ... one way that could be useful for her to recognize which line is ringing is by having a different ring tone for each. If BT100 cannot do it .. which phone can? Or is there some alternative way of helping the receptionist in this situation distinguish between the two lines? (Flash Operator Panel would not work well since she would not have it on all the time) Thanks, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.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] IP Softphone Recommendations
Can anyone recommend any free IP SoftPhones that are maybe open source? ___ Asterisk-Users mailing list Asterisk-Users@lists.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: SV: [Asterisk-Users] Re: IPswitch: How to use speed dialing?
Ronald,
I am more than happy to give you the 3 suggestions, when you
appologise to the list. Yes getting things to work can be
frustrating, and sometimes the answers are not as helpful as you'd
like, but I do refuse to help people who get irate on a public list
Especially when the outburst is to those who spend hours creating
programs that help many many people, those people who have talent
beyond my wildest dreams.
Please remember all advice on here is of a volentary nature, a lot
from people who could earn their crust providing this advice for a
charge, they don't, they spend hours helping and most of the time we
get it working - together
Now, take a deep breath, do the gentlemanly thing and lets see if we
can fix your issue.
David
On 4/26/05, Ronald Wiplinger [EMAIL PROTECTED] wrote:
Robert Goodyear wrote:
On Apr 25, 2005, at 10:55 PM, Ronald Wiplinger wrote:
Hi Ronald,
What happens in your Asterisk box when you press the Speed Dial
number in
IPS?
Can we make it so that you FIRST answer below questions, please?
| | Let's try it together:
Ronald: wow. Take a breath before you torch a generous developer. IPS
works like a charm for me in every way.
Seriously,
/rg
For me it doesn't!!!
And IF it works so good for you, why you are not willing / able to fill
the simple three fields out for me, telling me what you think would work.
I ask this question now several times, and get only answers, like it
works, it works for me too
FOR ME IT DOES NOT WORK, ...
So how did yo make it that it works for you???
Questions below to be kindly filled out:
| | Let's try it together:
| 1. Open IPswitch
| 2. Open Extensions tab on top
| 3. Switch to the tab Speed Dials on the bottom
| 4. Fill in:
| Name: [EMAIL PROTECTED]
| Caller Id: Peter
| Visible on Panel: (ticket)
| Exentension Group: Speed Dial Numbers
CLI answers:
| | | Congratualtions, you have successfully installed the Asterisk Open
| Source .
| tgj wrote:
|
|
|
| Hi Ronald,
|
| I must admit I am getting confused now.
|
| I understand that you have a problem getting Speed Dial Buttons to
| work.
| The problem as I understand it is that the calls are placed in the
| wrong
| context.
|
| To solve that problem I have asked you to make sure that you have
typed
| a
| valid context on the configuration page. Have you tried that?
|
| I think thats all you need to do, how do I post an example of that?
| It's a
| fairly easy thing to do.
|
| Thorben
|
|
|
|
|
| What is the right syntax to do that?
| Context for dialing a trunk line is trunkint
| Peter has the phone number 011-234-5678
| How to set it up as a speed dial number? Below are all info you may
| need:
|
| The phone 601 (= Monitor extension) is a Sip phone,
|
| [general]
| context=default; Default context for incoming calls
|
| [601]
| type=friend
| username=601
| secret=dont+tell+you
| canreinvite=no
| host=dynamic
| dtmfmode=rfc2833
| [EMAIL PROTECTED]
| nat=yes
| callgroup=1
| pickupgroup=1
| callerid=Ronald Hotline,601
| qualify=1000
|
|
| extensions.conf
| [default]
| ...
| include = trunkint
| ...
|
| [trunkint]
| ;
| ; International long distance through trunk
| ; . other lines deleted
| exten = _9011Z.,107,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
| exten = _9011Z.,108,hangup
|
|
|
|
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[Asterisk-Users] ACD in Asterisk
Hi everybody, I am having a problem while setting up queues in Asterisk. Callers are kept in the queues and told to wait while there are available agents. Even if I use ringall as strategy the call is not always sent to all free agents. Is there a problem with Automatic Call Distribution in Asterisk or am I missing something? Below is my queues.conf. Thanks for any suggestion Lamine [general] [default] [sceclient] music=sceclient strategy=leastrecent timeout=30 retry=5 wrapuptime=0 maxlen=0 announce-holdtime=no member=Agent/3001 member=IAX2/3000 member=IAX2/3001 ___ Asterisk-Users mailing list Asterisk-Users@lists.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] IP Softphone Recommendations
Best I have used is fireflyJ Dinesh. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William C. Lohr Jr. Sent: Tuesday, April 26, 2005 4:51 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] IP Softphone Recommendations Can anyone recommend any free IP SoftPhones that are maybe open source? ___ Asterisk-Users mailing list Asterisk-Users@lists.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] routing in extensions.conf
Thanks Stefan, you rule...
now, tell me just one more thing please,
I putted in capi.conf :
msn=12345678
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=siemens
devices=2
and in extension.conf :
[siemens]
exten = 930,1,Dial(SIP/joao)
but this means that when 930 is dialed, user joao always receives the
calls,
but I have 10 SIP users , and I whant that, after 930 have been dialed,
to dial one more number to refer to each of the SIP users. How do I put
it in extensions.conf ?
Thanks
Joao
Stefan Helbing wrote:
Hello Joao,
first I suggest you set an context string in capi.conf to lead incoming calls
into a special context to give you more flexibility (in my opinion), e.g.
context=siemens
For this you need a line [siemens] in your extensions.conf.
Then (and also in the case you use the default context for everything) you need the necessary lines in extensions.conf.
If you call the number 930 from siemens to asterisk you need a line like
exten = 930,1,DoWhatEverYouWantToDo
This line currently is missing therefor the fallback of asterisk to an s extensions. If you want to catch this, too (what I would recommend), you need an additional line
exten = s,1,DoStandardThings
Of course, this is only the minimum, there are much more possibilities
(especially if you want to do more than one thing in an extension).
Bye
Stefan
sth==Originalnachricht==
sthVon: Joao Pereira [EMAIL PROTECTED]
sthDatum: 2005-04-22 18:25:17
sthAn: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
sthBetreff: [Asterisk-Users] routing in extensions.conf
sth
sthHello all,
sthIm using chan_capi to connect from a Siemens High Path to a Aterisk,
sthwhen I call from the Asterisk clients to the Siemens PBX, it works, when
sthI call from a Siemens client to a SIP(Asterisk) client, it doesnt work
sthand says this:
sth
sth == Starting CAPI[contr1/930]/1 at default,930,1 failed so falling back
sthto exten 's'
sth == Starting CAPI[contr1/930]/1 at default,s,1 still failed so falling
sthback to context 'default'
sthApr 22 16:50:27 WARNING[1149236544]: pbx.c:1877 ast_pbx_run: Channel
sth'CAPI[contr1/930]/1' sent into invalid extension 's' in context
sth'default', but no invalid handler
sth
sthI think the problem is in the extensions.conf configuration, when the
sthSiemens calls the Asterisk, it starts ringing and nothing happens, but
sthwhat do I have to put in the extensions.conf to route the calls to the
sthcorrect SIP user?
sthThanks
sthJoao
sth
sth***
sthhere s my capi.conf
sth
sth[general]
sthnationalprefix=0
sthinternationalprefix=00
sthrxgain=0.8
sthtxgain=0.8
sth
sth[interfaces]
sthmsn=12345678
sthincomingmsn=*
sthcontroller=1
sthsoftdtmf=1
sthaccountcode=
sthcontext=default
sth;echosquelch=1
sth;echocancel=yes
sthdevices=2
sth
sth
sth***
sthhere s my extensions.conf
sth
sth[general]
sthstatic=yes
sthwriteprotect=no
sth
sth[globals]
sthCONSOLE=Console/dsp ; Console interface for demo
sthTRUNK=CAPI
sth
sth[default]
sth
sth; SIP to SIP
sthexten = 100,1,Dial(SIP/joao)
sthexten = 101,1,Dial(SIP/encoder)
sth
sth;SIP to Siemens
sthexten = 118,1,Dial,CAPI/12345678:b${EXTEN}|30
sthexten = 136,1,Dial,CAPI/12345678:b${EXTEN}|30
sthexten = 139,1,Dial,CAPI/12345678:b${EXTEN}|30
sthexten = 116,1,Dial,CAPI/12345678:b${EXTEN}|30
sthexten = 908,1,Dial,CAPI/12345678:b${EXTEN}|30
sth
sth;Siemens to SIP
sth;exten = s,1,Dial(SIP/joao) this one works, but it always dial the SIP
sthuser joao
sth
sthexten = 555,1,Dial(SIP/joao) ; ok, this is the one that doesnt work,
sthhow can I route the calls?
sth
sth
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sthhttp://lists.digium.com/mailman/listinfo/asterisk-users
sthTo UNSUBSCRIBE or update options visit:
sth http://lists.digium.com/mailman/listinfo/asterisk-users
sth
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[Asterisk-Users] Unexpected control subclass 17
Hi ! Starting this morning I start seeing this in my /var/log/asterisk/messages : Apr 26 08:52:13 WARNING[18321] file.c: Unexpected control subclass '17' Apr 26 08:52:57 WARNING[18321] file.c: Unexpected control subclass '17' Apr 26 11:21:17 WARNING[20842] file.c: Unexpected control subclass '17' Apr 26 11:21:37 WARNING[20842] file.c: Unexpected control subclass '17' Apr 26 11:22:14 WARNING[20842] file.c: Unexpected control subclass '17' What does it mean ? Should I do something about it ? TIA Peter De Schrijver ___ Asterisk-Users mailing list Asterisk-Users@lists.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 best practices
Meetme is a great tool .. It can give you a great conference capabilities ... Also using Faxing will be great ... Mohamed Farid ,, -Original Message- From: Craig Simon [mailto:[EMAIL PROTECTED] Sent: Monday, April 25, 2005 4:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk best practices List, I have been using asterisk for a couple of weeks now, to support some Cisco 7960 and 7920 phones, and have been enjoying the learning experience. I have gotten the phone firmware upgraded, Broadvoice connectivity, basic dial plan, and voice mail working. However I am sure that there is more that I can do. So my question, what is the best feature of Asterisk, and how have you deployed it in your organization? What trick configuration have you come up with to do something really out of the box cool? If you can document it with come configuration samples, so much the better. Thanks in advance Craig Notice: This e-mail (including attachments) is confidential and is intended solely for the addressee. Unless you are the addressee, you may not read, copy, use or store this e-mail in any way, or permit others to. If you have received it in error, please contact Mediterranean Smart Cards Company :+202 333 1400 ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Group/Broadcast Voicemail
Has anyone set up Group/Broadcast Voicemail for 50 or more mailboxes? ___ Asterisk-Users mailing list Asterisk-Users@lists.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] VOIP Gateways Asterisk
On 4/26/05, Callum McGillivray [EMAIL PROTECTED] wrote: My problem is that this installation is most likely to occur prior to the release of the new card (and definitely prior to it's vigorous testing in the field). If anyone can give me ideas at this point it would be appreciated. There are a number of cisco routers that will do the job for you but they are not cheap eg AS5350 8 E1's AS5400 16 E1's or other normal routers such as 3660's will support upto 4 E1'e Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.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] No busy tone when dialing out over ISDN with Polycom 500 IP
Then you'll need to check the value of DIALSTATUS and run Busy when needed. See extensions.conf.sample's [macro-stdexten]. Kib Eki wrote: Isn't the zapata.conf only for Digium hardware? I use an Eicon Card. Eric Wieling aka ManxPower wrote: Kib Eki wrote: Hi, what do i have to configure to get a busy tone when dialing out over ISDN channel with my Polycom 500 IP? Try priindication = inband in /etc/asterisk/zapata.con ___ Asterisk-Users mailing list Asterisk-Users@lists.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 capi: Long incomingmsn line in capi.conf?
On 4/26/05, Stefan Gofferje [EMAIL PROTECTED] wrote: Stefan Helbing schrieb: Hello, the incomingmsn line in chan_capi's capi.conf is limited to 80 characters (AST_MAX_EXTENSION default value). My problem: I have to include several MSNs but NOT all. The interface is a 30 channel PRI card with a number area of 600 numbers, splitted in different functions. Some numbers are used for fax, some for PPP, some for telephony. According to another email on this list, accept all incoming MSN's but create an entry in extensions.conf for each msn you wish to ignore (or wildcard) as follows exten = _123456XX,1,Wait(30) The wait will stop asterisk from answering the call so the other capi devices fax etc should then answer the call. Try it and let us know it would be good for future reference Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.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] bri cli error
Good day all I get this error in my cli chan_zap.c:7407 zt_pri_error: PRI: !! Got a UA, but i'm in state 0 I have a 4 port Junghannes card connect with 2 bri isdn lines It keeps on dropping calls and giving errors Please help and advice Thanks ALtus ___ Asterisk-Users mailing list Asterisk-Users@lists.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/NetMeeting
On 4/26/05, William C. Lohr Jr. [EMAIL PROTECTED] wrote: Does anyone know if it is possible to resolve an IP from outside a small LAN. I would like to be able to specify a SIP client that is outside my office LAN. The problem is that the isp will not provide a static IP that's affordable. I use a DYNDNS.org address with it. When I want to use NetMeeting for desktop sharing, I just ping the DYNDNS address and it gives me the current IP of the remote machine. Is it possible to specify the host name, say billscomputer.dyndns.org for the address of the SIP client in the appropriate .conf file for Asterisk? This is covered automatically if you set host=dynamic in sip.conf and have the sip phone register with your asterisk then asterisk knows what IP address the phone is on, this will be updated with every registration request. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Cisco Call Manager
Hi,
I'm integrating cisco call manager with asterisk
this is what I have in sip.conf
[callman]
type=friend
nat=no
insecure=very
context=dialplan
host=172.16.4.82
port=5060
disallow=all
allow=ulaw
allow=alaw
canreinvite=yes
qualify=yes
and this is my dial statement
Exten = _881.,1,Dial(sip/callman/${EXTEN})
when I call 88109 (that's handled by callman) I get
Executing Dial(SIP/88411-1cac, sip/callman/88109)
-- Called callman/88109
-- Got SIP response 503 Service Unavailable back from 172.16.4.82
-- SIP/callman-d037 is circuit-busy
If I call a non existant call manager extention I get
Executing Dial(SIP/88411-553a, sip/callman/88188)
-- Called callman/88188
-- Got SIP response 404 Not Found back from 172.16.4.82
-- SIP/callman-7371 is circuit-busy
Any idea of what is happening ?
I dont have access to callman logs, so I can only report what is
happening on my side.
--
Best regards,
Alessio mailto:[EMAIL PROTECTED]
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RE: [Asterisk-Users] Realtime voicemail
You entered 'other' as the context in the voicemail_users Database, but
you failed to specify that context when you made the call for voicemail
from the dial plan. The dial plan should be:
VoiceMail(SIP/601-a9a3, [EMAIL PROTECTED])
As stated in other posts on this subject, the Voicemail application
assumes the context to be 'default' unless otherwise specified.
Hope this helps.
Joe
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Terry
Wilson
Sent: Friday, April 08, 2005 1:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime voicemail
Try having the voicemail command do Voicemail([EMAIL PROTECTED])
On Apr 4, 2005 4:56 AM, Ronald Wiplinger [EMAIL PROTECTED] wrote:
I tried to use ONE entry of my voicemail.conf to put into the
database:
[other]
;602=1357,Ronald Wiplinger 602,[EMAIL PROTECTED]
INSERT INTO `voicemail_users` ( `uniqueid` , `customer_id` , `context`
, `mailbox` , `password` , `fullname` , `email` , `pager` , `stamp` ,
`attach` , `saycid` , `hidefromdir` ) VALUES ('1', '602', 'other',
'602', '3525', 'Ronald Wiplinger', '[EMAIL PROTECTED]', '',
NOW( ) , 'no', 'yes', 'no')
extconfig.conf includes:
voicemail = mysql,astconf,voicemail_users
*CLI reload
-- Executing VoiceMail(SIP/601-a9a3, b602) in new stack Apr 4
17:48:34 WARNING[18977]: app_voicemail.c:2227 leave_voicemail: No
entry in voicemail config file for '602'
What do I miss?
bye
Ronald
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[Asterisk-Users] i need Asterisk free Billing systems
Hello Is thereAsterisk free billing system?if you know a good system please reply me with it. thanks in advance Regards ___ Asterisk-Users mailing list Asterisk-Users@lists.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] ForkCDR question
hi if receiving a call, I lookup wheather or not that call should be diverted at user's request. if it should be diverted, I want to use ForkCDR to keep this entry orig srcorig dst and add one more orig dstdivert dst but with forkcdr, I only get orig dstorig dst can someone help me out here, please? roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to set jitter buffer for SIP
there is jitter setting for h323 in oh323.conf where to set min/max jitter buffer for SIP ? i am getting bad voice via *, maybe this jitter buffer setting will 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] Good FXO for UK use.
Title: Message Hi all, looking for some advice on a good FXO solution for Asterisk, living in the UK brings some issues (as Asterisk and associated hardwareappears to begeared to the US/Canadian market) such as calling line ID, impedance settings and echo cancellation (as a result of the impedance issues). I originally went down the route of an X100P which after applying the UK CLID patch worked fine (good sound levels and quality) except the echo issue. I decided then to follow the Sipura SPA-3000 route as this theoritically answered all my problems, unfortuntately the sound quality is appalling (see - http://voxilla.com/index.php?name=PNphpBB2file=viewtopict=474postdays=0postorder=aschighlight=3000+qualitystart=0). So I guess the obvious choice would be the TDM400 with FXO daughter board, I assume this works with my current zaptel drivers and UK CLID patch? Can anyone confirm this works fine in the UK or are there other suggestions? Regrads, Ray ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to set jitter buffer for SIP
there is jitter setting for h323 in oh323.conf where to set min/max jitter buffer for SIP ? i am getting bad voice via *, maybe this jitter buffer setting will 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] VOIP Gateways Asterisk
We do some least cost routing and pass calls to a Cisco AS5300 and are looking at also using a Lucent TNT. It's just a matter of setting up the dial plan in extensions.conf The cisco is using h323 and if the client does SIP the protocol conversion is done on the * box, otherwise it is basicaly just a hand off. John Dunham John Dunham CTO - Global Technologies Communications 713-559-9002 713-559-0001 FAX (234) 803-6732195 Nigeria GSM 832-922-9123 Cell ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Good FXO for UK use.
Date: Tue, 26 Apr 2005 12:36:26 +0100 From: Razza [EMAIL PROTECTED] Subject: [Asterisk-Users] Good FXO for UK use. To: asterisk-users@lists.digium.com Message-ID: !~!UENERkVCMDkAAQACABgAT1LIZaT+QEqJAR K3kpBGu8KQMG2M/[EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hi all, looking for some advice on a good FXO solution for Asterisk, living in the UK brings some issues (as Asterisk and associated hardware appears to be geared to the US/Canadian market) such as calling line ID, impedance settings and echo cancellation (as a result of the impedance issues). I originally went down the route of an X100P which after applying the UK CLID patch worked fine (good sound levels and quality) except the echo issue. I decided then to follow the Sipura SPA-3000 route as this theoritically answered all my problems, unfortuntately the sound quality is appalling (see - http://voxilla.com/index.php?name=PNphpBB2 http://voxilla.com/index.php?name=PNphpBB2file=viewtopict=4 74postday s=0postorder=aschighlight=3000+qualitystart=0 file=viewtopict=474postdays=0postorder=aschighlight=3000+ qualityst art=0). So I guess the obvious choice would be the TDM400 with FXO daughter board, I assume this works with my current zaptel drivers and UK CLID patch? Can anyone confirm this works fine in the UK or are there other suggestions? No the TDM400 does not work, it does not detect calling party termination correctly, so IVR and voicemail do not see the caller hang up on BT lines. Digium are aware of the problem, but fixing it doesn't seem to be a high priority, despite the fact that they have been supplied with detailed technical information regarding BT line behaviour :-(. FWIW, I find the Sipura 3000 works well for me here in the UK once configured using their on-line web tool. Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Good FXO for UK use.
Title: RE: [Asterisk-Users] Good FXO for UK use. So I guess the obvious choice would be the TDM400 with FXO daughter board, I assume this works with my current zaptel drivers and UK CLID patch? Can anyone confirm this works fine in the UK or are there other suggestions? Regrads, Ray The TDM400 works fine in the UK with an FXO daughterboard, the echo is eliminated although my callers do sometimes report that the line is a little quiet. I have noticed this on calling in but it is not major and a lot of it is possibly to do with the lack of quality on my BT line as a previous non VOIP switchboard plugged into the line was equally quiet (although with a hardwired phone it seems fine). R's Ian ___ Asterisk-Users mailing list Asterisk-Users@lists.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] pridialplan/TON question
Hi! I'm trying to understand how asterisk handles the TON (using the pridialplan=... directive). Setting the TON for outgoing calls using pridialplan and prilocaldialplan works fine. But how can I query and process the TON for incoming calls? e.g. in the follwing scenario: PBX--- asterisk PSTN 1. The PBX sends SETUP messages with the appropriate TON. I want to rewrite the called number into a common format to make an ENUM lookup. Thus, I need to query the TON sent by the PBX and add the correct prefixes. 2. Further, in case of unsuccessful ENUM lookups, I want to forward the SETUP message to the PSTN, again using the appropriate TON. CVS version allows the setting of pridialplan=dynamic. But I want to use stable as this is for a stable machine. Can I implement this with stable asterisk? thanks, Klaus ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Good FXO for UK use.
Title: RE: [Asterisk-Users] Good FXO for UK use. No the TDM400 does not work, it does not detect calling party termination correctly, so IVR and voicemail do not see the caller hang up on BT lines. Digium are aware of the problem, but fixing it doesn't seem to be a high priority, despite the fact that they have been supplied with detailed technical information regarding BT line behaviour :-(. Patrick is right about this , I get 20 or so seconds of solid tone at the end of all my voicemails, but I can live with this for the sake of no echo. R's Ian ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Shanghai or Bangalore DIDs
Hello, does someone offer DIDs from the areas of shanghai and/or bangalore. Many thanks, Marc -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- 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] pridialplan/TON question
Klaus Darilion wrote: Hi! I'm trying to understand how asterisk handles the TON (using the pridialplan=... directive). Setting the TON for outgoing calls using pridialplan and prilocaldialplan works fine. But how can I query and process the TON for incoming calls? e.g. in the follwing scenario: PBX--- asterisk PSTN 1. The PBX sends SETUP messages with the appropriate TON. I want to rewrite the called number into a common format to make an ENUM lookup. Thus, I need to query the TON sent by the PBX and add the correct prefixes. 2. Further, in case of unsuccessful ENUM lookups, I want to forward the SETUP message to the PSTN, again using the appropriate TON. CVS version allows the setting of pridialplan=dynamic. But I want to use stable as this is for a stable machine. Can I implement this with stable asterisk? I always thought that if you set pridialplan=unknown the telco would not munge the digits. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] pridialplan/TON question
Eric Wieling aka ManxPower wrote: Klaus Darilion wrote: ... e.g. in the follwing scenario: PBX--- asterisk PSTN 1. The PBX sends SETUP messages with the appropriate TON. I want to rewrite the called number into a common format to make an ENUM lookup. Thus, I need to query the TON sent by the PBX and add the correct prefixes. 2. Further, in case of unsuccessful ENUM lookups, I want to forward the SETUP message to the PSTN, again using the appropriate TON. CVS version allows the setting of pridialplan=dynamic. But I want to use stable as this is for a stable machine. Can I implement this with stable asterisk? I always thought that if you set pridialplan=unknown the telco would not munge the digits. Anyway, if I set TON to unknown, I have to send the number according to the PSTN dialing plan (00 for int, 0 for national, ...). Thus, if the PBX does not use UNKNOWN, I have to translate the numbers out of their original TON to ton=unknown. Therefore, I need to process the incoming TON. How do I handle this? regards, klaus ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Cisco Systems to Acquire Sipura Technology
SAN JOSE, Calif., April 26, 2005 - Cisco Systems® today announced a definitive agreement to acquire privately-held Sipura Technology, Inc. This represents Cisco's first acquisition for its Linksys division, the leading provider of wireless and networking hardware for home, Small Office/Home Office (SOHO) and small business environments. Sipura is a leader in consumer voice over internet protocol (VoIP) technology and is a key technology provider for Linksys' current line of VoIP networking devices. In addition to Sipura's valuable technology and customer relationships, their experienced team with extensive VoIP expertise will help build a foundation for Linksys' internal research and development capabilities in voice, video and other markets. Full Story: http://newsroom.cisco.com/dlls/2005/corp_042605.html?DCMP=BAC-TS01 ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Alternatives to SpanDSP??
Why would you expect a bunch of fax modems to work any better than spandsp? If spandsp doesn't work reliably your system is very likely broken. I have had hundreds of complaints about spandsp reliability. I have analysed at least 50 or 60 audio logs. I have found maybe 5 or 6 which has real spandsp problems. The rest had frame slips. Of the 5 or 6 with real problems, most have been fixed in the latest version. I have one weird audio log from a new HP combination printer and fax machine that i haven't sorted out yet. These HP machines really are total crap. I have workarounds in spandsp for several blatently wrong things they do. I don't yet know who is at fault with this latest problem. Regards, Steve Jeremy Melanson wrote: More like, I already have enough Digium cards, and I don't want purchase a bunch of fax/modems and more Digium cards than I alrady have. I have a PRI line that I'd like to support high-volume faxing on. I've gotten SpanDSP to work with * over the PRI, but I need a more reliability. That, and I guess I'm probably just being cheap too :-) - Jeremy On Mon, 2005-04-25 at 13:15 -0500, Anton Krall wrote: Maybe I started the day slow :) but let me see if I undertood correctly. You say that you don't want to rely on having to buy Digums or any other type of cards in oder to tie everything into spandsp and * but you would rather have dedicated PSTN lines with faxes on them? |-Original Message- |From: [EMAIL PROTECTED] | |I guess I didn't word this right. |It's not that SpanDSP ties up extensions, as it definitely |doesn't. I was more referring to the standard hardware-based |solutions out there that need to have a dedicated line for an |incoming fax. I need the ability to send and receive faxes |with a good amount of reliability, and would love to integrate |it with Asterisk. I'm just not keen on needing to buy a bunch |of Digium TDM cards just to support such a solution. | |Don't get me wrong, SpanDSP is great! I'm just looking for |something a little more enterprise-ready. | |On Mon, 2005-04-25 at 12:07 -0500, Eric Wieling aka ManxPower wrote: | I wasn't aware that SpanDSP tied up a bunch of extensions. | | Jeremy Melanson wrote: |I'm trying to see if anyone knows of an alternative solution, | commercial or non-commercial, to SpanDSP. I'm specifically looking |for another software-based, DSP fax that doesn't require me to add a tie up a | bunch of extensions on my PBX. | | Has anyone ever seen such an animal, or gotten such it to play nice | with Asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Softphone Recommendations
Can anyone recommend any free IP SoftPhones that are maybe open source? Mine is not open source, but it's free for non-commercial use. Give it a try http://www.marccharbonneau.com/asterisk/mediaxphone.php ___ Asterisk-Users mailing list Asterisk-Users@lists.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 and Cisco Call Manager
try with
type=peer
good luck
Edgar
On Tue, 2005-04-26 at 13:08 +0200, Alessio Focardi wrote:
Hi,
I'm integrating cisco call manager with asterisk
this is what I have in sip.conf
[callman]
type=friend
nat=no
insecure=very
context=dialplan
host=172.16.4.82
port=5060
disallow=all
allow=ulaw
allow=alaw
canreinvite=yes
qualify=yes
and this is my dial statement
Exten = _881.,1,Dial(sip/callman/${EXTEN})
when I call 88109 (that's handled by callman) I get
Executing Dial(SIP/88411-1cac, sip/callman/88109)
-- Called callman/88109
-- Got SIP response 503 Service Unavailable back from 172.16.4.82
-- SIP/callman-d037 is circuit-busy
If I call a non existant call manager extention I get
Executing Dial(SIP/88411-553a, sip/callman/88188)
-- Called callman/88188
-- Got SIP response 404 Not Found back from 172.16.4.82
-- SIP/callman-7371 is circuit-busy
Any idea of what is happening ?
I dont have access to callman logs, so I can only report what is
happening on my side.
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Re: [Asterisk-Users] VOIP Gateways Asterisk
You might look into the Lucent TNT's, they do SIP/MGCP (with the Hash codes, os 10.1.xx+), and also terminate modem calls. They are cheap (check ebay, www.qualitek.net) and their are loads of them out there. One TNT will handle your requirements easily, their is an example on the wiki on how to use a TNT with asterisk, and it works properly. I used Asterisk to talk to them via SIP, didn't try mgcp but it should work fine. Regards Michael Baird Hi all, I was just wondering if someone could help me with info on VOIP Gateways. We are planning to do an * install in an apartment building, this building is going to require somewhere in the vacinity of 20 E1 lines (each with 30 voice channels). Short of buying 20 Servers with Digium cards, what are my options in having the E1 lines terminate on some other hardware and then having the calls passed through to Asterisk to perform the PBX type functionality ? I have heard that using some form of VOIP gateway should help, but I really have no idea how this works. Any help would be appreciated. Cheers, Callum ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Digium for ETSI ISDN
Hi, I just wanted to know if Digium support ETSI ISDN? Thanks. Cheers, Angelo ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Shanghai or Bangalore DIDs
I'm also looking for numbers from HongKong, Taiwan, Japan and Singapore So if someone has some DIDs from this areas, I'm very interested to get one or another from those DIDs. Best Regards, Marc Marc Storck wrote: Hello, does someone offer DIDs from the areas of shanghai and/or bangalore. Many thanks, Marc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- 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] Digium for ETSI ISDN
Hi Nathaniel, ETSI ISDN is used by 99% of the world's ISDN E1s, so you can guess the answer. :-) ETSI ISDN is also known as CTR4, Net5 and most commonly EuroISDN. It is known as EuroISDN in the * config files. Regards, Steve Nathaniel Angelo A. Torres (247talk) wrote: Hi, I just wanted to know if Digium support ETSI ISDN? Thanks. Cheers, Angelo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial CLI Command
I have just installed a release version 1.0.7 of asterisk: I already installed in past asterisk and in my previous installation I may find the dial command on CLI that now I haven't found: it is possible? The lack of dial CLI command is an upgrade?or Is there some problem in my installation? Thanks 4 help me now f. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Digium for ETSI ISDN
Thanks Steve. Cheers, Angelo -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Tuesday, April 26, 2005 8:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Digium for ETSI ISDN Hi Nathaniel, ETSI ISDN is used by 99% of the world's ISDN E1s, so you can guess the answer. :-) ETSI ISDN is also known as CTR4, Net5 and most commonly EuroISDN. It is known as EuroISDN in the * config files. Regards, Steve Nathaniel Angelo A. Torres (247talk) wrote: Hi, I just wanted to know if Digium support ETSI ISDN? Thanks. Cheers, Angelo ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Alternatives to SpanDSP??
On April 26, 2005 08:12 am, Steve Underwood wrote: Why would you expect a bunch of fax modems to work any better than spandsp? If spandsp doesn't work reliably your system is very likely broken. I've had spandsp crash out on some kind of floating point error about a half dozen times over about 250 faxes When it crashes it takes Asterisk down with it. These systems are SuperMicro Xeon server-class systems, no overclocking, RAM was tested overnight with memtest86, no-nonsense, nothing funny type machines. SpanDSP and Asterisk were both compiled with the same compiler without any oddball optimizations (just whatever's in the default makefiles). It's a bitch to try and reconstruct, but it's the only reason I'm not using spandsp in production; when I was using spandsp I had it on a completely separate machine on the local LAN to avoid the spandsp crashes from taking the voice part down. -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] asterisk xlite nat problem
Dear All, I am new to this mailing list , I have bought some digium cards to play with , Installed it and configured asterisk . I was able to test voicemail IVR , I succeeded also to use xlite from a windows machine to call another phone through a PSTN line. and call the xlite client from a PSTN line. All these tests was made while Xlite was in the same LAN as asterisk . The problem evolves when I try to use xlite to connect to asterisk from the Internet . The xlite client is using dial-up connection.Direct real IP. while the asterisk client is behind a NAT router but I am using a default server setting that forwards any incoming traffic to the asterisk machine. the problem asterisk can initiate a call to xlite while xlite times out and fails to connect to asterisk. given that any other raffic can reach from the xlite machine to the asterisk machine like ssh tftp ping TCP, UDP, ICMP but the traffic from xlite didn't reach the asterisk machine at all. even on port 5060. Any help of guidance is appreciated. -- Mostafa Ibrahim Technical Team Leader Linux-Plus Information Systems www.linux-plus.com Maadi, Cairo, Egypt. Cornich El-Nil Tel : +202-5276616 : +202-5240745 Fax : +202-5261055 ___ Asterisk-Users mailing list Asterisk-Users@lists.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] pridialplan/TON question
On Tue, 26 Apr 2005, Klaus Darilion wrote: Anyway, if I set TON to unknown, I have to send the number according to the PSTN dialing plan (00 for int, 0 for national, ...). Thus, if the PBX does not use UNKNOWN, I have to translate the numbers out of their original TON to ton=unknown. Therefore, I need to process the incoming TON. How do I handle this? You have two options: 1) Use the CALLINGTON variable in the dialplan. This is only for the calling party number, not the called party number. 2) Use the internationalprefix, nationalprefix, localprefix etc settings in the zapata.conf file. I _think_ this will affect both the interpretation of calling and called party and possibly also the TON of the called number for outgoing links. I am nut sure under which circumstances these variables are applied. Isdn handling in Asterisk tends to be these kinds of hacks. 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] Alternatives to SpanDSP??
Hi Steve, I sent a mail to this list a week ago regarding exactly this issue. Spandsp doesn't work for me (getting 200rows tiffs), but sending and receiving faxes through a FXS-FXO bridge (a TDM11B) works without problems. My motherboard is based an Aopen AK33 (VIA686a chipset, KT133, 700Mhz Athlon). I've disabled USB, 2nd IDE, VGA interrupt (runing without X), sound.. I've also tweaked PCI settings in the BIOS, testing each time, but I don't know what can be wrong. Here is some more info: cat /proc/interrupts CPU0 0: 23411526 XT-PIC timer 2: 0 XT-PIC cascade 4: 80 XT-PIC serial 8: 1 XT-PIC rtc 10: 23322936 XT-PIC wctdm 12: 1 XT-PIC acpi 14: 91663 XT-PIC ide0 15: 51573 XT-PIC eth0 NMI: 0 ERR: 0 $ ./zttest Opened pseudo zap interface, measuring accuracy... 99.975586% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% Thanks Julian J. M. On 4/26/05, Steve Underwood [EMAIL PROTECTED] wrote: Why would you expect a bunch of fax modems to work any better than spandsp? If spandsp doesn't work reliably your system is very likely broken. I have had hundreds of complaints about spandsp reliability. I have analysed at least 50 or 60 audio logs. I have found maybe 5 or 6 which has real spandsp problems. The rest had frame slips. Of the 5 or 6 with real problems, most have been fixed in the latest version. I have one weird audio log from a new HP combination printer and fax machine that i haven't sorted out yet. These HP machines really are total crap. I have workarounds in spandsp for several blatently wrong things they do. I don't yet know who is at fault with this latest problem. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial CLI Command
I have just installed a release version 1.0.7 of asterisk: I already installed in past asterisk and in my previous installation I may find the dial command on CLI that now I haven't found: it is possible? The lack of dial CLI command is an upgrade?or Is there some problem in my installation? Thanks 4 help me now f. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Alternatives to SpanDSP??
I'd have to second this, it works flawlessly for us, the issues we do have are with devices not properly turning off echo cancellation... W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Tuesday, April 26, 2005 7:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Alternatives to SpanDSP?? Why would you expect a bunch of fax modems to work any better than spandsp? If spandsp doesn't work reliably your system is very likely broken. I have had hundreds of complaints about spandsp reliability. I have analysed at least 50 or 60 audio logs. I have found maybe 5 or 6 which has real spandsp problems. The rest had frame slips. Of the 5 or 6 with real problems, most have been fixed in the latest version. I have one weird audio log from a new HP combination printer and fax machine that i haven't sorted out yet. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Good FXO for UK use.
Title: Message This maybe an out of place comment but it would appear Digium show little to no interest in non-North Americanimplementations, do we know if they are ever going to resolve this issue? or indeed how much it would cost? Based on my experience I'm sure there are a number of UK based people who could jointly fund such a development for a reasonable FXO product? Patrick Lidstone wrote:No the TDM400 does not work, it does not detect calling party termination correctly, so IVR and voicemail do not see the caller hang up on BT lines. Digium are aware of the problem, but fixing it doesn't seem to be a high priority, despite the fact that they have been supplied with detailed technical information regarding BT line behaviour :-(. Ian D. Willoughbywrote :Patrick is right about this , I get 20 or so seconds of solid tone at the end of all my voicemails,but I can live with this for the sake of no echo. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Phone Recommendation.
yes *71 will disable call waiting on any phone used with [EMAIL PROTECTED] see the Handbook for more info http://asteriskathome.sourceforge.net/handbook/ --- Anton Krall [EMAIL PROTECTED] wrote: How? You mean if you use [EMAIL PROTECTED] right? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Wiley Siler |Sent: Lunes, 25 de Abril de 2005 02:50 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] Phone Recommendation. | |Call waiting can be disabled in Asterisk via *71 regardless of |the phone used. | |Cheers, |Wiley | | | | |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of Sean A. |Newton |Sent: Monday, April 25, 2005 11:56 AM |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] Phone Recommendation. | | |I'm looking for recommendations for a office phone that has |the ability to disable call-waiting. | |Needs to be similar in features to a Polycom IP300. | |Thanks, | |--Sean | |-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- | sean a. newton [EMAIL PROTECTED] | louisville, ky, usa http://wewt.net | | Another day, another convertible and another hotel | full of cops.-- Hunter S. Thompson |-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pridialplan/TON question
Anyway, if I set TON to unknown, I have to send the number according to the PSTN dialing plan (00 for int, 0 for national, ...). Thus, if the PBX does not use UNKNOWN, I have to translate the numbers out of their original TON to ton=unknown. Therefore, I need to process the incoming TON. How do I handle this? You have two options: 1) Use the CALLINGTON variable in the dialplan. This is only for the calling party number, not the called party number. 2) Use the internationalprefix, nationalprefix, localprefix etc settings in the zapata.conf file. I _think_ this will affect both the interpretation of calling and called party and possibly also the TON of the called number for outgoing links. I am nut sure under which circumstances these variables are applied. Isdn handling in Asterisk tends to be these kinds of hacks. I have a Digium E100P card, with an EuroISDN PRI E1. On incoming calls the CALLINGTON variable is empty. I have the latest stable version of asterisk. Do I have to use another variable or is the TON only support in CVS? Marc -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- 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] Alternatives to SpanDSP??
Hi Andrew, If you can catch one of these events, and get a traceback of the stack, I will take a look. This is not happening to most users, so it must be some specific combination of things on your machine. I have reports of high volume faxing running for extended periods from some users. Regards, Steve Andrew Kohlsmith wrote: On April 26, 2005 08:12 am, Steve Underwood wrote: Why would you expect a bunch of fax modems to work any better than spandsp? If spandsp doesn't work reliably your system is very likely broken. I've had spandsp crash out on some kind of floating point error about a half dozen times over about 250 faxes When it crashes it takes Asterisk down with it. These systems are SuperMicro Xeon server-class systems, no overclocking, RAM was tested overnight with memtest86, no-nonsense, nothing funny type machines. SpanDSP and Asterisk were both compiled with the same compiler without any oddball optimizations (just whatever's in the default makefiles). It's a bitch to try and reconstruct, but it's the only reason I'm not using spandsp in production; when I was using spandsp I had it on a completely separate machine on the local LAN to avoid the spandsp crashes from taking the voice part down. -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] Alternatives to SpanDSP??
Hi Julian, Sounds like a frame slip problem if the result depends on the source. Most people, including me, have trouble with the TDM cards. They worked without problem when I was first developing the FAX software in spandsp, so I assume the TDM driver has gathered bugs since that time. Regards, Steve Julian J. M. wrote: Hi Steve, I sent a mail to this list a week ago regarding exactly this issue. Spandsp doesn't work for me (getting 200rows tiffs), but sending and receiving faxes through a FXS-FXO bridge (a TDM11B) works without problems. My motherboard is based an Aopen AK33 (VIA686a chipset, KT133, 700Mhz Athlon). I've disabled USB, 2nd IDE, VGA interrupt (runing without X), sound.. I've also tweaked PCI settings in the BIOS, testing each time, but I don't know what can be wrong. Here is some more info: cat /proc/interrupts CPU0 0: 23411526 XT-PIC timer 2: 0 XT-PIC cascade 4: 80 XT-PIC serial 8: 1 XT-PIC rtc 10: 23322936 XT-PIC wctdm 12: 1 XT-PIC acpi 14: 91663 XT-PIC ide0 15: 51573 XT-PIC eth0 NMI: 0 ERR: 0 $ ./zttest Opened pseudo zap interface, measuring accuracy... 99.975586% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% Thanks Julian J. M. On 4/26/05, Steve Underwood [EMAIL PROTECTED] wrote: Why would you expect a bunch of fax modems to work any better than spandsp? If spandsp doesn't work reliably your system is very likely broken. I have had hundreds of complaints about spandsp reliability. I have analysed at least 50 or 60 audio logs. I have found maybe 5 or 6 which has real spandsp problems. The rest had frame slips. Of the 5 or 6 with real problems, most have been fixed in the latest version. I have one weird audio log from a new HP combination printer and fax machine that i haven't sorted out yet. These HP machines really are total crap. I have workarounds in spandsp for several blatently wrong things they do. I don't yet know who is at fault with this latest problem. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP4000 Conference Phone
I was afraid you would say that. Does anyone out there have the latest firmware for the Soundpoint IP 4000? Thanks, Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul HalesSent: Monday, April 25, 2005 7:45 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Polycom IP4000 Conference Phone You need to have a very new firmware... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley SilerSent: Tuesday, 26 April 2005 6:33 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Polycom IP4000 Conference Phone Can someone verify that this phone uses the same configs and sip.ld and other files as the IP 500 ? I jus tgot one and I cannot get it provisioned yet. Thanks, Wiley CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Phone Recommendation.
Yep. Or if you hand code the feature into your dial plan too W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Monday, April 25, 2005 6:13 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Phone Recommendation. How? You mean if you use [EMAIL PROTECTED] right? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of Wiley |Siler |Sent: Lunes, 25 de Abril de 2005 02:50 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] Phone Recommendation. | |Call waiting can be disabled in Asterisk via *71 regardless of the |phone used. | |Cheers, |Wiley | | | | |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of Sean A. |Newton |Sent: Monday, April 25, 2005 11:56 AM |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] Phone Recommendation. | | |I'm looking for recommendations for a office phone that has the ability |to disable call-waiting. | |Needs to be similar in features to a Polycom IP300. | |Thanks, | |--Sean | |-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- | sean a. newton [EMAIL PROTECTED] | louisville, ky, usa http://wewt.net | | Another day, another convertible and another hotel | full of cops.-- Hunter S. Thompson |-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.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] Incoming Not Answering
Hello I have just setup my first Asterisk box and Im having a great time. I am having a little trouble getting incoming calls to answer. This is what I see on the console: Apr 25 17:01:07 NOTICE[3514]: chan_zap.c:5374 ss_thread: Got event 2 (Ring/Answered)... -- Executing Wait(Zap/1-1, 1) in new stack -- Executing Answer(Zap/1-1, ) in new stack -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Apr 25 17:01:15 NOTICE[3515]: chan_zap.c:5374 ss_thread: Got event 2 (Ring/Answered)... -- Executing Wait(Zap/1-1, 1) in new stack -- Executing Answer(Zap/1-1, ) in new stack -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' I have the demo config files in place which show the s extension being answered and a message played but this is not happening. Any assistance is appreciated. Thank you, Dave ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Alternatives to SpanDSP??
As Steve has mentioned several times, it seems the TDM-fxo boards have an issue with missed frames that no one is addressing. Very few (if any) TDM users have been able to make spandsp function correctly, and the few that might have it working don't know why. Having played around some with zttest (modifying the code to better understand the issues), it would appear the TDM card consumes about 1.02 seconds to obtain one second of data. That would suggest the card misses about one frame in every fifty. Haven't figured out why as yet and don't know that I've got the practical experience to actually find the root cause. I sent a mail to this list a week ago regarding exactly this issue. Spandsp doesn't work for me (getting 200rows tiffs), but sending and receiving faxes through a FXS-FXO bridge (a TDM11B) works without problems. My motherboard is based an Aopen AK33 (VIA686a chipset, KT133, 700Mhz Athlon). I've disabled USB, 2nd IDE, VGA interrupt (runing without X), sound.. I've also tweaked PCI settings in the BIOS, testing each time, but I don't know what can be wrong. Here is some more info: cat /proc/interrupts CPU0 0: 23411526 XT-PIC timer 2: 0 XT-PIC cascade 4: 80 XT-PIC serial 8: 1 XT-PIC rtc 10: 23322936 XT-PIC wctdm 12: 1 XT-PIC acpi 14: 91663 XT-PIC ide0 15: 51573 XT-PIC eth0 NMI: 0 ERR: 0 $ ./zttest Opened pseudo zap interface, measuring accuracy... 99.975586% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% Thanks Julian J. M. On 4/26/05, Steve Underwood [EMAIL PROTECTED] wrote: Why would you expect a bunch of fax modems to work any better than spandsp? If spandsp doesn't work reliably your system is very likely broken. I have had hundreds of complaints about spandsp reliability. I have analysed at least 50 or 60 audio logs. I have found maybe 5 or 6 which has real spandsp problems. The rest had frame slips. Of the 5 or 6 with real problems, most have been fixed in the latest version. I have one weird audio log from a new HP combination printer and fax machine that i haven't sorted out yet. These HP machines really are total crap. I have workarounds in spandsp for several blatently wrong things they do. I don't yet know who is at fault with this latest problem. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Alternatives to SpanDSP??
On Tue, 2005-04-26 at 08:57 -0400, Andrew Kohlsmith wrote: On April 26, 2005 08:12 am, Steve Underwood wrote: Why would you expect a bunch of fax modems to work any better than spandsp? If spandsp doesn't work reliably your system is very likely broken. I've had spandsp crash out on some kind of floating point error about a half dozen times over about 250 faxes When it crashes it takes Asterisk down with it. These systems are SuperMicro Xeon server-class systems, no overclocking, RAM was tested overnight with memtest86, no-nonsense, nothing funny type machines. SpanDSP and Asterisk were both compiled with the same compiler without any oddball optimizations (just whatever's in the default makefiles). It's a bitch to try and reconstruct, but it's the only reason I'm not using spandsp in production; when I was using spandsp I had it on a completely separate machine on the local LAN to avoid the spandsp crashes from taking the voice part down. I was under the impression that pretty much all of these problems were usually traced to the version of libtiff that was in use... Perhaps you should try to track it down/solve the problem rather than patch it over? Of course, it is sometimes difficult to keep working on solving a problem when you don't have the knowledge to find the problem, and a client just wants it to work right :) Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.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] return a value from dial macro
Does anyone know of a way to pass a value back to the dial plan after
calling a macro from the dial app in the 1.0 release? I think this
should be pretty simple, but I can't quite figure out how.
The example would work except that the modified value of found is not
usable when Dial ends. I think that the MACRO_RESULT would do this, but
it doesn't appear to have made it into 1.0
I want to stop going through the priorities after completion of a
successful dial, but only if MachineDetect returns 0. If it returns 1 I
want to hangup on the called party and goto the next priority
exten = 223,3,SetVar(__found=0)
exten = 223,4,Dial(SIP/[EMAIL PROTECTED],48,rtgM(md))
exten = 223,5,GotoIf($[${found} = 1]?7)
exten = 223,6,Voicemail(u${EXTEN})
exten = 223,7,Hangup
[macro-md]
exten = s,1,MachineDetect(700,2,2200)
exten = s,2,GotoIf($[${MACHINE} = 1]?3:5)
exten = s,3,SoftHangup(${CHANNEL})
exten = s,4,Goto(6)
exten = s,5,SetVar(found=1)
exten = s,6,NoOp
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[Asterisk-Users] Cisco to buy Sipura
If this has already been posted I apologize for the redundant post. http://newsroom.cisco.com/dlls/2005/corp_042605.html?DCMP=BAC-TS01 -- Cory Andrews Senior Partner VOIPSupply.com + V: 800.398.VOIP X22 F: 716.630.1548 E: [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] Alternatives to SpanDSP??
On Tue, 2005-04-26 at 14:11 +0100, Julian J. M. wrote: Hi Steve, I sent a mail to this list a week ago regarding exactly this issue. Spandsp doesn't work for me (getting 200rows tiffs), but sending and receiving faxes through a FXS-FXO bridge (a TDM11B) works without problems. My motherboard is based an Aopen AK33 (VIA686a chipset, KT133, 700Mhz Athlon). I've disabled USB, 2nd IDE, VGA interrupt (runing without X), sound.. I've also tweaked PCI settings in the BIOS, testing each time, but I don't know what can be wrong. Here is some more info: I think the useful debug info is the audio files rxfax will record if you enable the debugging These would allow Steve to re-create what happened, and I assume, fix spandsp and perhaps even test it using the same input file... Personally, I'd like to see (and I assume so would Steve) everyone who has all the required debug info, send it to Steve so that we can end up with a better fax solution. In fact, I think we would probably end up being MORE compatible than any other fax product on the market (well, maybe :) Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.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] pridialplan/TON question
Hi Peter! Peter Svensson wrote: On Tue, 26 Apr 2005, Klaus Darilion wrote: Anyway, if I set TON to unknown, I have to send the number according to the PSTN dialing plan (00 for int, 0 for national, ...). Thus, if the PBX does not use UNKNOWN, I have to translate the numbers out of their original TON to ton=unknown. Therefore, I need to process the incoming TON. How do I handle this? You have two options: 1) Use the CALLINGTON variable in the dialplan. This is only for the calling party number, not the called party number. Bad thing. I guess this is an important feature when interacting with existing PBXs. How are other people deal with this (processing the TON of the called number)? 2) Use the internationalprefix, nationalprefix, localprefix etc settings in the zapata.conf file. I _think_ this will affect both the interpretation of calling and called party and possibly also the TON of the called number for outgoing links. I am nut sure under which circumstances these variables are applied. Nothing of this is included in stable version. I'm sure I'm not the first person putting an asterisk box between a PBX and the telco line. Is everboy using asterisk CVS out there? regards, klaus ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Phone Recommendation.
That's what I figured.. Im not using [EMAIL PROTECTED] ... Plain ol' good asterisk... |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |[EMAIL PROTECTED] |Sent: Martes, 26 de Abril de 2005 08:28 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] Phone Recommendation. | |yes *71 will disable call waiting on any phone used with |[EMAIL PROTECTED] | |see the Handbook for more info |http://asteriskathome.sourceforge.net/handbook/ | |--- Anton Krall [EMAIL PROTECTED] wrote: | How? You mean if you use [EMAIL PROTECTED] right? | | |-Original Message- | |From: [EMAIL PROTECTED] | |[mailto:[EMAIL PROTECTED] On | Behalf Of | |Wiley Siler | |Sent: Lunes, 25 de Abril de 2005 02:50 p.m. | |To: Asterisk Users Mailing List - Non-Commercial | Discussion | |Subject: RE: [Asterisk-Users] Phone Recommendation. | | | |Call waiting can be disabled in Asterisk via *71 | regardless of | |the phone used. | | | |Cheers, | |Wiley | | | | | | | | | |-Original Message- | |From: [EMAIL PROTECTED] | |[mailto:[EMAIL PROTECTED] On | Behalf Of Sean A. | |Newton | |Sent: Monday, April 25, 2005 11:56 AM | |To: asterisk-users@lists.digium.com | |Subject: [Asterisk-Users] Phone Recommendation. | | | | | |I'm looking for recommendations for a office phone | that has | |the ability to disable call-waiting. | | | |Needs to be similar in features to a Polycom IP300. | | | | |Thanks, | | | |--Sean | | | ||-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- | | sean a. newton | [EMAIL PROTECTED] | | louisville, ky, usa | http://wewt.net | | | | Another day, another convertible and another hotel | | | full of cops.-- Hunter S. Thompson | ||-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- | | | |___ | |Asterisk-Users mailing list | |Asterisk-Users@lists.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 | | |__ |Do You Yahoo!? |Tired of spam? Yahoo! Mail has the best spam protection |around http://mail.yahoo.com |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pridialplan/TON question
Marc Storck wrote: Anyway, if I set TON to unknown, I have to send the number according to the PSTN dialing plan (00 for int, 0 for national, ...). Thus, if the PBX does not use UNKNOWN, I have to translate the numbers out of their original TON to ton=unknown. Therefore, I need to process the incoming TON. How do I handle this? You have two options: 1) Use the CALLINGTON variable in the dialplan. This is only for the calling party number, not the called party number. 2) Use the internationalprefix, nationalprefix, localprefix etc settingsin the zapata.conf file. I _think_ this will affect both theinterpretation of calling and called party and possibly also theTON of the called number for outgoing links. I am nut sure underwhich circumstances these variables are applied. Isdn handling in Asterisk tends to be these kinds of hacks. I have a Digium E100P card, with an EuroISDN PRI E1. On incoming calls the CALLINGTON variable is empty. I have the latest stable version of asterisk. Do I have to use another variable or is the TON only support in CVS? CALLINGTON is only in CVS. internationalprefix, ... is only in CVS or using stable patched with bristuff. regards, klaus ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Digium for ETSI ISDN
On 4/26/05, Nathaniel Angelo A. Torres (247talk) [EMAIL PROTECTED] wrote: Hi, I just wanted to know if Digium support ETSI ISDN? Yes ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Fail over solutions
Hi folks, I'm curious; What does everyone do for failover? I have two servers, same os/compilation. I designate one the master, the other the slave, and I rsync the config files once an hour and trigger a restart when convenient command on the console. These two servers are setup in the dns in a round robin fashion. What is everyone else doing? Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.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] YAC and IPs
Guys. Im using YAC to send callerid info to PCs and I was wondering if there is a way to get the IP of a certain SIP or IAX client/technology when a dial command is issued. For example, if the dialplan has a dial sip/client or iax2/client, is there a way to get the current clients IP so I can pass the parameters to the system call that send the YAC callerid info? ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Zap/PRI: received AOC-E charging
Trying to make a call via our PRI: (CVS everything, CVS-NHEAD-04/23/05-16:08:12) -- Executing Dial(IAX2/[EMAIL PROTECTED], Zap/R2/2815699900|30) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called R2/2815699900 -- Channel 0/19, span 2 got hangup -- Channel 0/19, span 2 received AOC-E charging 0 units Apr 26 09:06:49 WARNING[10040]: chan_zap.c:7457 zt_pri_error: PRI: Call Reference Length not supported: 0 -- Zap/43-1 is circuit-busy -- Hungup 'Zap/43-1' Any idea on what AOC-E means? Here is a full pri debug: -- Executing Dial(IAX2/[EMAIL PROTECTED], Zap/R2/2815699900|30) in new stack -- Making new call for cr 32775 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=93 Call Ref: len= 2 (reference 7/0x7) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 92] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 18 ] [1c 23 9f aa 06 80 01 00 82 01 00 8b 01 00 a1 15 02 01 0c 02 01 00 80 0d 4d 61 74 74 68 65 77 20 42 6f 65 68 6d] Facility (len=37, codeset=0) [ 0x9f, 0xaa, 0x06, 0x80, 0x01, 0x00, 0x82, 0x01, 0x00, 0x8b, 0x01, 0x00, 0xa1, 0x15, 0x02, 0x01, 0x0c, 0x02, 0x01, 0x00, 0x80, 0x0d, 'Matthew', 0x20, 'Boehm' ] [1e 02 80 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [28 0e b1 4d 61 74 74 68 65 77 20 42 6f 65 68 6d] Display (len=14) Charset: 31 [ Matthew Boehm ] [6c 06 21 80 33 30 34 34] Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '3044' ] [70 0b a1 32 38 31 35 36 39 39 39 30 30] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '2815699900' ] -- Called R2/2815699900 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 7/0x7) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 82 a9] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Temporary failure (41), class = Network Congestion (2) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/18, span 2 got hangup -- Channel 0/18, span 2 received AOC-E charging 0 units Protocol Discriminator: Q.931 (8) len=7 Apr 26 09:08:12 WARNING[10040]: chan_zap.c:7457 zt_pri_error: PRI: Call Reference Length not supported: 0 Apr 26 09:08:12 WARNING[10040]: chan_zap.c:7457 zt_pri_error: PRI: Call Reference Length not supported: 0 Call Ref: len= 0 (reference 0/0x0) (Originator) Message type: RELEASE COMPLETE (90) [08 02 82 a9] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Temporary failure (41), class = Network Congestion (2) ] Apr 26 09:08:12 WARNING[10040]: chan_zap.c:7457 zt_pri_error: PRI: Call Reference Length not supported: 0 -- Making new call for cr 0 -- Processing IE 8 (cs0, Cause) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Zap/42-1 is circuit-busy NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/42-1' == Everyone is busy/congested at this time (1:0/1/0) -- Hungup 'IAX2/[EMAIL PROTECTED]' Thanks for help, Matthew -- Matthew Boehm, IT DirectorCypress Telecommunications [EMAIL PROTECTED] 3838 N. Sam Houston Parkway E #400 T: 832-200-8640 x3044 Houston, TX 77032 My girlfriend was recently diagnosed with multiple personality disorder; When she called yesterday, my CallerID box exploded. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] pridialplan/TON question
On Tue, 26 Apr 2005, Klaus Darilion wrote: You have two options: 1) Use the CALLINGTON variable in the dialplan. This is only for the calling party number, not the called party number. Bad thing. I guess this is an important feature when interacting with existing PBXs. How are other people deal with this (processing the TON of the called number)? It would not be very hard to create a CALLEDTON variable. The information is sent from libpri to chan_zap. A few more fields in chan_zap and a little bit of code in pbx.c. I really _ really_ wish asterisk would stop using the pseudo-variables and simply store stuff in the dialplan variables (like PRI_CAUSE etc already do). These pseudo-variables are stupid since in most cases reading and writing them is not time critical. 2) Use the internationalprefix, nationalprefix, localprefix etc settings in the zapata.conf file. I _think_ this will affect both the interpretation of calling and called party and possibly also the TON of the called number for outgoing links. I am nut sure under which circumstances these variables are applied. Nothing of this is included in stable version. I'm sure I'm not the first person putting an asterisk box between a PBX and the telco line. Is everboy using asterisk CVS out there? We use cvs from an old date (predating these functions) but with quite a few additional patches of our own. there is currenctly a showstopper bug where the dtmf-detecting dsp is disabled on outbound call legs. Bad if you need #-transfers. 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] VOIP Gateways Asterisk
On 4/26/05, Michael Baird [EMAIL PROTECTED] wrote: You might look into the Lucent TNT's, they do SIP/MGCP (with the Hash codes, os 10.1.xx+), and also terminate modem calls. They are cheap (check ebay, www.qualitek.net) and their are loads of them out there. One TNT will handle your requirements easily, their is an example on the wiki on how to use a TNT with asterisk, and it works properly. I used Asterisk to talk to them via SIP, didn't try mgcp but it should work fine. The cisco routers will do sip as well. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Incoming Not Answering
-Original Message- snipped From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Sampson -- Executing Answer(Zap/1-1, ) in new stack -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' Apr 25 17:01:15 NOTICE[3515]: chan_zap.c:5374 ss_thread: Got event 2 (Ring/Answered)... -- Executing Wait(Zap/1-1, 1) in new stack -- Executing Answer(Zap/1-1, ) in new stack -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (default, s, 3) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' === Hi David, Can you post your extensions.conf file, there may be a clue somewhere in the exten = s, section. If you included the default example it should be working, but there may be something that has changed. Chad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP behind IPTables/NAT
Hi All, Can anyone help me out here? I'm having some issues configuring my IPTables firewall to properly NAT SIP and RTP packets to my asterisk server hiding behind it. Here are my current rules: #Inbound SIP to HERMES $IPTABLES -A PREROUTING -t nat -i $EXTIF -p udp --dport 5060 -j DNAT --to 192.168.123.4:5060 $IPTABLES -A FORWARD -i $EXTIF -p udp -d 192.168.123.4 --dport 5060 -j ACCEPT #Inbound RTP to HERMES $IPTABLES -A PREROUTING -t nat -i $EXTIF -p udp --dport 1:2 -j DNAT --to 192.168.123.4:1:2 $IPTABLES -A FORWARD -i $EXTIF -p udp -d 192.168.123.4 --dport 1:2 -j ACCEPT When I dial out via my SIP provider I appear to get a partial connection (the phone rings... that's a good sign) but no audio. Inbound I just get a busy and asterisk sees nothing. SIP SHOW REGISTRY shows me as registered with the remote host. Something else that worries me is that I'm seeing the good old Attempting native bridge... message when the destination picks up which, to my understanding, shouldn't happen since I have canreinvite=no set for both my SIP phone and SIP provider. Make sense to anyone? Ian ___ Asterisk-Users mailing list Asterisk-Users@lists.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]I wanted to understand
Hello all, I am new to Asterisk and I tried a basic configuration with 2 SIP phones (FCI IP Ranger), and it works very well !! I wanted to understand furthermore the asterisk product, his technical architecture, and after that try to understand how to add functionnalities thanks to te API. I have read that there are existing APIs, but I didn't find any documentation on how to use them ! Can someone direct me to some URLs ? Or tell me how to try to begin to understand the code ? Thank you very much for advance for your responses ! ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Alternatives to SpanDSP??
On Tue, 2005-04-26 at 08:34 -0600, Rich Adamson wrote: As Steve has mentioned several times, it seems the TDM-fxo boards have an issue with missed frames that no one is addressing. Very few (if any) TDM users have been able to make spandsp function correctly, and the few that might have it working don't know why. Having played around some with zttest (modifying the code to better understand the issues), it would appear the TDM card consumes about 1.02 seconds to obtain one second of data. That would suggest the card misses about one frame in every fifty. Haven't figured out why as yet and don't know that I've got the practical experience to actually find the root cause. Hmmm, interesting... my box is has a X100P, a TDM40B and a TE410p, and I don't seem to have a problem receiving a fax (via the TE410p) yet. ie, I haven't had any complaints, and maybe 10 successful faxes, so it isn't exactly foolproof, but so far so good. I'd still like to see someone say they receive some large number of faxes daily with spandsp from random senders (ie, not 100 faxes/day from the same junk fax sender :) Oh, and a description of their equipment would also be nice Even number of concurrent faxes they process, etc... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P + spandsp locks machine with zaptel asterisk 1.0.7
Hello, I've got an older machine that is being locked/hung by what appears to be the X100P card. I'm running the following: [9:[EMAIL PROTECTED]:[/home/jesse]# qpkg -v -I zaptel net-misc/zaptel-1.0.7 * [9:[EMAIL PROTECTED]:[/home/jesse]# qpkg -v -I asterisk net-misc/asterisk-oh323-0.6.5 * net-misc/asterisk-app_rtxfax-0.0.2_pre10 * net-misc/asterisk-1.0.7 * [9:[EMAIL PROTECTED]:[/home/jesse]# qpkg -v -I spandsp media-libs/spandsp-0.0.2_pre10 * # uname -a Linux rhea 2.6.11-gentoo-r2 #1 SMP Mon Mar 21 15:28:42 EST 2005 i686 Pentium II(Deschutes) GenuineIntel GNU/Linux It runs fine for about a day, then it locks solid overnight. I'm using the X100P as an answering machine and fax machine. The asterisk machine also serves SIP over my DSL line. When the machine locks up, if I connect a monitor and keyboard to it, I get no signal. The fans are running, but it would appear that no interrupts are being serviced. No SSH, no video, nothing. The X100P shares a line with the rest of the phones in my house, including the DSL modem. I have a line filter on every piece of equipment in the house, including my satellite receiver. The machine seems to hang overnight. Never during the day. My satellite dials out at 3am over the phone line to send billing info and retrieve guide updates. Is there a possibility that the modem tones generated by the satellite are hanging the zaptel kernel modules in my asterisk server? Is anyone else experiencing regular system lockups while running spandsp or zaptel? -- Jesse Guardiani, Systems Administrator WingNET Internet Services, P.O. Box 2605 // Cleveland, TN 37320-2605 423-559-LINK (v) 423-559-5145 (f) http://www.wingnet.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] japanese voice files
Anybody would have the japanese voice files for *? I need now the number's recording at least. Thanks, Isamar ___ Asterisk-Users mailing list Asterisk-Users@lists.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] IP Softphone Recommendations
Time Bandit wrote: Can anyone recommend any free IP SoftPhones that are maybe open source? Mine is not open source, but it's free for non-commercial use. Give it a try http://www.marccharbonneau.com/asterisk/mediaxphone.php I´m using X-lite on windows and linux, looks pretty well. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Alternatives to SpanDSP??
On April 26, 2005 09:47 am, Adam Goryachev wrote: I was under the impression that pretty much all of these problems were usually traced to the version of libtiff that was in use... Perhaps you should try to track it down/solve the problem rather than patch it over? Nope; it's not a tiff issue; using the clean (source-built) libtiff recommended by spandsp (3.5.7 I think offhand?) -- it was failing inside of spandsp with the FPU exception. I think I posted about it here before, let me see if I can dig it up. Of course, it is sometimes difficult to keep working on solving a problem when you don't have the knowledge to find the problem, and a client just wants it to work right :) :-) Well in this case I'm my own client, but I have 35 people in the same office who tend to raise holy hell when things like the phones don't work. :-) -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] Good FXO for UK use.
Title: Message I just bought a DigitNetworks card called "DigitNetworks X100P - FXO PCI card" which supposedly is compatible with the discontinued Digium X100P card. This is a single port FXO card. Tell me how to test forthe TDM400 problem and I'll perform a test and post my results back to the list. The card is dead cheap $25 (but $36 :-( for shipping ). Regards Johan. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RazzaSent: 26 April 2005 14:25To: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Good FXO for UK use. This maybe an out of place comment but it would appear Digium show little to no interest in non-North Americanimplementations, do we know if they are ever going to resolve this issue? or indeed how much it would cost? Based on my experience I'm sure there are a number of UK based people who could jointly fund such a development for a reasonable FXO product? Patrick Lidstone wrote:No the TDM400 does not work, it does not detect calling party termination correctly, so IVR and voicemail do not see the caller hang up on BT lines. Digium are aware of the problem, but fixing it doesn't seem to be a high priority, despite the fact that they have been supplied with detailed technical information regarding BT line behaviour :-(. Ian D. Willoughbywrote :Patrick is right about this , I get 20 or so seconds of solid tone at the end of all my voicemails,but I can live with this for the sake of no echo. ** Please note: The e-mail accompanying this disclaimer is confidential and may also be privileged. Please notify us immediately if you are not the intended recipient. You should not copy it, forward it, or use it for any purpose or disclose the contents to any person. This email has been swept for viruses using tools from our preferred suppliers. Telamon Systems actively supply both mail-scanning and anti-virus products in addition to supplying a range of security, infrastructure and business solutions to our customers. For further details please see our web site at www.telamon.co.uk, email [EMAIL PROTECTED] or call our sales team on +44 (0)870 607 4747 ** ___ Asterisk-Users mailing list Asterisk-Users@lists.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] CLI dial command
I have just installed a release version 1.0.7 of asterisk: I already installed in past asterisk and in my previous installation I may find the dial command on CLI that now I haven't found: it is possible? The lack of dial CLI command is an upgrade?or Is there some problem in my installation? Thanks 4 help me now f. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Alternatives to SpanDSP??
Having played around some with zttest (modifying the code to better understand the issues), it would appear the TDM card consumes about 1.02 seconds to obtain one second of data. That would suggest the I would like to chime in with my experience: We are trying to use SpanDSP off of a PRI to recieve 200 faxes a day. It's gone OK but not perfect. Some gotchas that I have found: 1. Timing (as others have said) is totally critical. I found a subtle timing error because our Asterisk box is behind an Adtran channel bank and the Adtran introduced tiny slips occasionally. Upgrading the firmware in the adtran and monkeying around with how the Adtran took it's timing from the PRI took care of it (after consultation with Adtran tech support which is first-rate BTW) 2. ZTTEST is a critical metric. I was getting disconnects on about 20% of faxes until I looked at the output of ZTTEST and found that it was dropping below 99.98% occasionally. Using setpci I changed the latency on the Zaptel boards (T100P TDM04) to the max, 254 and cranked down the latency on everything else as low as I dared. Now, I get 99.9873% across the board as long as I run the test, and I even get the magic 100% on 1 in 10 test passes. 3. Yes, we have the HP problem, and I don't know how I'm going to deal with it yet. I'll probably set up a problem fax line with an analog fax and give that number to those people that have the problem. It's always the same guys. I'm getting a reject rate of about 2-3% which is ok but the endusers of course want no rejects. I have to offset that with the convenience of getting the faxes as PDF's (we would take the paper fax and scan it into our CRM if you can believe it) and the monetary savings of not printing the faxes; we have a click rate from our print vendor and he loves it when we make paper 'cause it's more money for him. No more busy signals on the fax line is a bonus too, people being people the fax will sit idle all day then 15-20 faxes will try to come in simultaneously. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] YAC and IPs
On Tuesday 26 April 2005 9:58 am, Anton Krall wrote: Guys. Im using YAC to send callerid info to PCs and I was wondering if there is a way to get the IP of a certain SIP or IAX client/technology when a dial command is issued. For example, if the dialplan has a dial sip/client or iax2/client, is there a way to get the current clients IP so I can pass the parameters to the system call that send the YAC callerid info? Simplest way probably would be to parse the output of 'sip show peers' or a similar IAX CLI command (I dont use IAX, so I dunno.) I've got a small perl script that parses 'sip show peers' to get the peer name (SIP/whatever) and the matching IP address - just a simple regex exercise, really. It could easily be converted to AGI where one could call: exten = s,1,AGI(tech2ip.pl|SIP/whatever) ; IP is put in TECH_IP If anybody wants the script, let me know. Cheers! -josiah ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Alternatives to SpanDSP??
We have terrible problems sending faxes via the TDM cards. Not even using SpanDSP. Just TE110P for the telco side and TDM400P for the fax machine. Steve Underwood wrote: Hi Julian, Sounds like a frame slip problem if the result depends on the source. Most people, including me, have trouble with the TDM cards. They worked without problem when I was first developing the FAX software in spandsp, so I assume the TDM driver has gathered bugs since that time. Regards, Steve Julian J. M. wrote: Hi Steve, I sent a mail to this list a week ago regarding exactly this issue. Spandsp doesn't work for me (getting 200rows tiffs), but sending and receiving faxes through a FXS-FXO bridge (a TDM11B) works without problems. My motherboard is based an Aopen AK33 (VIA686a chipset, KT133, 700Mhz Athlon). I've disabled USB, 2nd IDE, VGA interrupt (runing without X), sound.. I've also tweaked PCI settings in the BIOS, testing each time, but I don't know what can be wrong. Here is some more info: cat /proc/interrupts CPU0 0: 23411526 XT-PIC timer 2: 0 XT-PIC cascade 4: 80 XT-PIC serial 8: 1 XT-PIC rtc 10: 23322936 XT-PIC wctdm 12: 1 XT-PIC acpi 14: 91663 XT-PIC ide0 15: 51573 XT-PIC eth0 NMI: 0 ERR: 0 $ ./zttest Opened pseudo zap interface, measuring accuracy... 99.975586% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% Thanks Julian J. M. On 4/26/05, Steve Underwood [EMAIL PROTECTED] wrote: Why would you expect a bunch of fax modems to work any better than spandsp? If spandsp doesn't work reliably your system is very likely broken. I have had hundreds of complaints about spandsp reliability. I have analysed at least 50 or 60 audio logs. I have found maybe 5 or 6 which has real spandsp problems. The rest had frame slips. Of the 5 or 6 with real problems, most have been fixed in the latest version. I have one weird audio log from a new HP combination printer and fax machine that i haven't sorted out yet. These HP machines really are total crap. I have workarounds in spandsp for several blatently wrong things they do. I don't yet know who is at fault with this latest problem. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pridialplan/TON question
On Tue, 26 Apr 2005, Marc Storck wrote: I have a Digium E100P card, with an EuroISDN PRI E1. On incoming calls the CALLINGTON variable is empty. I have the latest stable version of asterisk. Do I have to use another variable or is the TON only support in CVS? CALLINGTON was not populated in -stable. Tha patch was only added to -head. It is not that hard to add, I can send you our old patch if you want it. 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] Phone Recommendation.
On Mon, 25 Apr 2005, Wiley Siler wrote: Call waiting can be disabled in Asterisk via *71 regardless of the phone used. Cheers, Wiley Well, this is part of a larger problem I'm having. I can't get CheckGroup/SetGroup to work as I think it should for my dynamically added ACD agents. The management here is frustrated, and they just want to buy a few phones that simply can have call waiting disabled. --Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
