Re: [asterisk-users] E1 not coming up
here's my output of /proc/zaptel/4: Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 HDB3/CCS/CRC4 94 TE4/0/4/1 Clear (In use) 95 TE4/0/4/2 Clear (In use) 96 TE4/0/4/3 Clear (In use) 97 TE4/0/4/4 Clear (In use) 98 TE4/0/4/5 Clear (In use) 99 TE4/0/4/6 Clear (In use) 100 TE4/0/4/7 Clear (In use) 101 TE4/0/4/8 Clear (In use) 102 TE4/0/4/9 Clear (In use) 103 TE4/0/4/10 Clear (In use) 104 TE4/0/4/11 Clear (In use) 105 TE4/0/4/12 Clear (In use) 106 TE4/0/4/13 Clear (In use) 107 TE4/0/4/14 Clear (In use) 108 TE4/0/4/15 Clear (In use) 109 TE4/0/4/16 HDLCFCS (In use) 110 TE4/0/4/17 Clear (In use) 111 TE4/0/4/18 Clear (In use) 112 TE4/0/4/19 Clear (In use) 113 TE4/0/4/20 Clear (In use) 114 TE4/0/4/21 Clear (In use) 115 TE4/0/4/22 Clear (In use) 116 TE4/0/4/23 Clear (In use) 117 TE4/0/4/24 Clear (In use) 118 TE4/0/4/25 Clear (In use) 119 TE4/0/4/26 Clear (In use) 120 TE4/0/4/27 Clear (In use) 121 TE4/0/4/28 Clear (In use) 122 TE4/0/4/29 Clear (In use) 123 TE4/0/4/30 Clear (In use) 124 TE4/0/4/31 Clear (In use) i get this output for all other working spans aswell. What is the output of `cat /proc/zaptel/spannumberthatsbroken` --- Matthew Fredrickson Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7941 localized menus with SIP firmware
If you flash new sip flash firmware into 7941 look at tftp log, you will see, that after firmware flashing and phone reboot, it will download and flash localization files in next flashing cycle, if you copy this files from callmanager tftp dir to your tftp server it will work. before flasing localized menus, you will need to edit .cnf.xml config file, look at tags like user/networklocale localized phone menus has nothing to do with sccp, if you use sip firmware on phone. PJ Greg Oliver wrote: On Tue, 2007-06-26 at 21:45 +0200, Olivier wrote: Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards Actually Cisco only sendx xml for certain things. It uses a modified SIP stack and it's native SCCP stack to provision button templates, softkeys, etc.. I did hours of packet captures to try and get the info, but it is embedded into the call control stack of their phones. If you read the chan_sccp code a bit, it has a few different button layout options, that are encoded in the SCCP driver and not xml files. I wish they would go to all config files, but I doubt they will... -Greg ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Kirk IP600 V3 DECT Wireless server
Hi Remco I have used the IP600 v3 with SIP support on Asterisk... apparently I was the 1st person globally to run it at a site. The 1st firmware was a bit buggy at times, but seems to be much better on the later versions. Kind Regards Garth Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO MSN:[EMAIL PROTECTED] Web:www.bitco.co.za Remco Barendse wrote: Hi list! Does anyone have experiences with the updated model of the Kirk IP600? The V3 model is supposed to support SIP instead of only SCCP or H323 which would make the use with Asterisk a lot easier. I have only tested the Kirk IP600 V2 with SCCP / Skinny protocol which is still giving me severe headaches : - the standard Skinny driver in * doesn't work, only the version of Sergio Chersovani is working but with some stability problems - audible echo - i have 2 kirk handsets and 2 Siemens handsets, the first 2 Kirk handsets always ring on an incoming call, the Siemens handsets only sometimes ring (don't know if this is caused by a problem in the SCCP driver or poor support of the IP600 for non-Kirk phones??) Because of my earlier experiences I am hesitating to try the V3 version. Experiences anyone? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How many number of parallel calls can make through asterisk
http://www.voip-info.org/wiki/view/Asterisk+dimensioning Santosh S Kumar wrote: Hi, We are planning to develop a product making asterisk as base, I love that asterisk is open source and eager to start working on it. But before even we get into start working on asterisk we want to know how many number of parallel calls can be made from a single asterisk box, considering we install the latest stable version of asterisk (we are ready to buy the enterprise version if there is any) on a highly configured box. So, how many number of parallel calls can we make through asterisk?? Regards, -- thanks, Yusuf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How many number of parallel calls can make through asterisk
Hi, We are planning to develop a product making asterisk as base, I love that asterisk is open source and eager to start working on it. But before even we get into start working on asterisk we want to know how many number of parallel calls can be made from a single asterisk box, considering we install the latest stable version of asterisk (we are ready to buy the enterprise version if there is any) on a highly configured box. So, how many number of parallel calls can we make through asterisk?? Regards, -- S Santosh Kumar BlueCornea Network, http://surabisantosh.blogspot.com http://www.bluecornea.com/services PhoneNo. 91 0 9886243523 [ http://surabisantosh.blogspot.com/2007/05/what-name-can-you-come-up-with-be.html ] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk call unique id in dialplan
Hi
how can i retrieve the call unique id of asterisk in the dialplan?
I have enabled the cdr logging on a postgres database.
In the table cdr each record has a field that assumes an unique id
(for example: 1141628669.51)
Can i retrieve this from the dialplan?
For example:
exten = 203,1,Answer
exten = 203,2,Set(CALLERID(name)=UNIQUE_ID - ${var_name_unique_id})
exten = 203,3,Dial(SIP/203)
Can i do something similar that?
How can i retrieve the unique_id generated?
thanks.
--
/*/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser
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[asterisk-users] nway call
I'm using asterisk 1.4.5 , on Centos. My kernel version is 2.6.9-55.ELsmp I've configured the nway call. I made entries in extension.conf, feature.conf, as per required. I'm trying to make a 3-way conference with the 1 user myself ( using asterisk), and two others are PSTN line users. I'm making a first call , then putting that person on hold by pressing **( as per feature.conf entry), then dialing the third pasrt number, at this time the first person( second party) is listening the music on hold. The next number is geeting connect but i m not able to take back that person in conference by pressing the required keys *0. Any if any of you have made that then please let me know. Regards, Kesh - Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Call Wainting dysfunctions
I am trying to implement a Centralized Call Waiting System. I have red
some document about asterisk group features to manage group and
category of a sip channel.
I have done a lot of test about it but always it doesn't work
correctly if I transfer the call.
This is the macro code I use for inbound calls.
[macro-test]
; ${ARG1} - technology something like SIP
; ${ARG2} - resource. snom300-for-vasya
; ${ARG3} - dial timeout
; ${ARG4} - dial options
; ${ARG5} - dial url
exten = s,1,Goto(s-set-variables,1)
exten = s,n(set_var_ret),Set(GROUP(${LOCAL_PARTY})=OUTBOUND_GROUP)
exten = s,n,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])}
1]?play_back_busy)
exten = s,n,GotoIf($[${LEN(${CALLERIDNUM})} != 4]?skip_down)
exten = s,n,Set(GROUP(${CALLERIDNUM})=OUTBOUND_GROUP)
exten = s,n(skip_down),Noop(Test)
exten = s,n,NoOp(${OUTBOUND_GROUP})
exten = s,n,Dial(${TECHNOLOGY}/${LOCAL_PARTY}|${DIAL_TIMEOUT})
exten = s,n(macro_out),MacroExit()
exten = s,n(play_back_busy),GotoIf($[zoo${BUSYOPT} = zooNoBusy]?macro_out)
exten = s,n,Busy
; // set variables
exten = s-set-variables,1,Set(TECHNOLOGY=${ARG1})
exten = s-set-variables,n,Set(LOCAL_PARTY=${ARG2})
exten = s-set-variables,n,Set(DIAL_TIMEOUT=${ARG3})
exten = s-set-variables,n,Goto(s,set_var_ret)
; // check dnd
I have only SIP hardware Phone with g729 codec.
When I call an internal from another one this is the output of group
show channels in console:
ChannelGroup Category
SIP/3673-b73ef7c0 OUTBOUND_GROUP3673
SIP/3673-b73ef7c0 OUTBOUND_GROUP3671
When I am tranfering the call from the destination to another
extension this is the output of the same command:
ChannelGroup Category
SIP/3671-b73501c0 OUTBOUND_GROUP3671
SIP/3671-b73501c0 OUTBOUND_GROUP3700
SIP/3673-b73ef7c0 OUTBOUND_GROUP3673
SIP/3673-b73ef7c0 OUTBOUND_GROUP3671
After the transfer the output is the follow but the call now is
between 3673 and 3700 extensions.
ChannelGroup Category
SIP/3673-b73ef7c0 OUTBOUND_GROUP3673
SIP/3673-b73ef7c0 OUTBOUND_GROUP3671
The extension 3700 seems free but it is busy.
The extension 3671 seems busy but it is free.
How can I implement a really working centralized call waiting feature?
Thanx a lot.
Lorenzo
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Re: [asterisk-users] Linking Asterisk with another SIP PBX (or SIP Softswitch)
Hi Noah; The reason that I am asking wether I need to determine the context is what I read in the documentation (about configuring outbound IAX connections), it did not mention the context at all, please read the below paragraph (I copy it from the documentation and paste it): Configuring Outbound IAX Connections: While an IAX user receives inbound calls; an IAX peer is used to place outbound calls. This section will set up iax.conf and extensions.conf so that you can place calls. iax.conf Configuration: The following entry in iax.conf can be used to place a call on the FWD network: [iaxfwd] type=peer host=iax2.fwdnet.net username=fwd-account-number secret=fwd-account-password qualify=yes disallow=all allow=ulaw allow=gsm allow=ilbc allow=g726 As you see that no context written in the iaxfwd peer configuration, so how it will work? Is it because the type is peer? Regards Bilal If I need to do a trunk between Asterisk and another SIP softswitch (so Asterisk will send a SIP calls to that softswitch), then I have to configure this on the sip.conf file Yes. And is it the same when I configure iax trunk? Not exactly the same, but very close. Here's a page on how to connect Asterisk to Cisco Call Manager using SIP: http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration Should I determine the context in this case for this SIP trunk? Yes. - Noah Need a vacation? Get great deals to amazing places on Yahoo! Travel. http://travel.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Query
Hi, I am trying to establish call through sip phone between two PC connected to linux box on which asterisk server is running 1st PC is having IP Adress : 192.168.1.149 2nd PC is having IP Adress : 192.168.1.53 I am trying to dial from 1st PC to 2nd PC through asterisk server The problem is 1st PC is calling directly to 2nd PC not through asterisk server I am doing the following additions in configuration files 1) sip.conf [general] context=sip bindport=5060 bindaddr=0.0.0.0 [phone1] type=friend host=192.168.1.149 port=5060 nat=yes dtmfmode=rfc2833 context=sip [phone2] type=friend host=192.168.1.53 port=5060 nat=yes dtmfmode=rfc2833 context=sip 2) extensions.conf exten = 11,1,Dial(SIP/phone2,20,tr) Now, I am calling from sip phone1 by name [EMAIL PROTECTED] It is not being called through asterisk server running on linux m/c. as no packet dumping us taking palce. As, I am running sip debub no messages are seen on screen. What additional routing informations are to be added to sip.conf, inorder to make it work . Thanx and regards sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MOH question w/Cisco 79xx phones
Hi Everyone Got a newbie type question regarding MOH Cisco phones. I'm still new to Asterisk (very new in fact) built up a AsteriskNOW box just to get something going. My simple test system has just 3 Cisco phones a 7905, 7940 7960. - Everything's running SIP. The 3 phones can call each other fine. - Can even leave (and retreive) voicemail messages. - No problems. My problem is with MOH (Music On Hold). - The 7940/7960 are somewhat okay, but the 7905 is another story. When a is call from a 7940/7960 is placed on 'hold' (by the calling party), MOH starts up on the 7940/7960, plays for about a second or two, then drops out for about a second or so, then continues. - After that, it continues to play okay. But when a call from the 7905 is placed on 'hold' (by the calling party), MOH starts up on the 7905, plays for a second or two, drops out for a sec, starts again for a sec or so, drops out, starts back up, drops out, etc., etc., etc Just up and down. - Kinda' like a Yo-Yo. Also - When the call from the 7905 is placed on hold, I see the following warning at the Asterisk CLI: [Jun 29 22:18:28] NOTICE[3376]: rtp.c783 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 10.0.0.110 I don't see this warning when the 7940/7960 is playing MOH. I'm using basic default settings for just about everything. - Could this be with the RTP config? - The 7905 Audio settings? Anybody have a clue? Thanks in advance. Gary Guthary ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX over T1
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andres Paglayan Sent: Friday, June 29, 2007 12:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX over T1 On Jun 22, 2007, at 3:43 PM, Joe acquisto wrote: I have an existing Hylafax system using a mainpine 4 port board, 4 POTS lines. Have a recently installed Asterisk system, with a dedicated T1 line. (Well, it's really a fonality system). What would I need to do, or where is the reading material, for what I need to do, to convert the Hylafax server to use the T1 line? Reliably. Preferably to use DID's as well. The current FAX works fine, but there is some desire to get rid of the analog lines. Could one add some sort of device in the Asterisk server, to act as FAX extensions, keeping the mainpine on the hylafax? Like a TDM400p with FSX modules? I'm just saying, ya know? I suppose I have to ask fonality, since it's their box? joe a. did you fix this yet? I had the same problem, and worked it out, contact me off list if you want the how to (or at least one of the how tos) Andres Paglayan --Harmony is more important than being right Bapak Why not post the how-tos to the list for anyone down the road that is searching for this same issue? Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH question w/Cisco 79xx phones
I think in your SIPDefault.cnf you disable VAD enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary Sent: Friday, June 29, 2007 9:28 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] MOH question w/Cisco 79xx phones Hi Everyone Got a newbie type question regarding MOH Cisco phones. I'm still new to Asterisk (very new in fact) built up a AsteriskNOW box just to get something going. My simple test system has just 3 Cisco phones a 7905, 7940 7960. - Everything's running SIP. The 3 phones can call each other fine. - Can even leave (and retreive) voicemail messages. - No problems. My problem is with MOH (Music On Hold). - The 7940/7960 are somewhat okay, but the 7905 is another story. When a is call from a 7940/7960 is placed on 'hold' (by the calling party), MOH starts up on the 7940/7960, plays for about a second or two, then drops out for about a second or so, then continues. - After that, it continues to play okay. But when a call from the 7905 is placed on 'hold' (by the calling party), MOH starts up on the 7905, plays for a second or two, drops out for a sec, starts again for a sec or so, drops out, starts back up, drops out, etc., etc., etc Just up and down. - Kinda' like a Yo-Yo. Also - When the call from the 7905 is placed on hold, I see the following warning at the Asterisk CLI: [Jun 29 22:18:28] NOTICE[3376]: rtp.c783 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 10.0.0.110 I don't see this warning when the 7940/7960 is playing MOH. I'm using basic default settings for just about everything. - Could this be with the RTP config? - The 7905 Audio settings? Anybody have a clue? Thanks in advance. Gary Guthary ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .call file
Nitesh Divecha wrote:
Hello All,
Is there any way to pass additional parameters while calling AGI from
*.call file?
Channel: Local/[EMAIL PROTECTED]
MaxRetries: 0
RetryTime: 15
WaitTime: 15
Application: AGI
Data: recordvoice.php
Something like Data: recordvoice.php?id=3453name=asterisk
Cheers,
Nitesh
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I'm not 100% sure if you can pass it directly, but you can use the set
option in the call file to set local variables within Asterisk and then
pass them to the AGI script. So for your example it would be.
Set: name=asterisk
This will set the variable ${name} in asterisk and depending how your
script was created you should be able to grab the variable to use within
the script. If you are using say the PHP AGI you can use something like
the following:
$var = $agi-get_variable(name);
This will create an array with $var['data'] holding 'asterisk';
Now one more thing I am not sure of is for multiple variables (haven't
tried it yet ;D ). You may have to do it one of two ways.
Set: name=asterisk, id=3453
or
Set: name=asterisk
Set: id=3453
and if those don't work, just format it so you can filter it out with PHP.
Hopefully this will help.
Kevin
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[asterisk-users] Asterisk 1.4 Warnnings
Dear Users ! I have recently installed asterisk 1.4 i got a warning message whenever i use reload or extensions reload. [Jun 29 19:22:11] WARNING[4539]: pbx.c:6236 ast_context_verify_includes: Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls' [Jun 29 19:22:11] WARNING[4539]: pbx.c:6236 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-canonical' [Jun 29 19:22:11] WARNING[4539]: pbx.c:6236 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-customers' [Jun 29 19:22:11] WARNING[4539]: pbx.c:6236 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-via-pstn' please help what is this and how i can fix it? Regards, Umar Draz _ Picture this share your photos and you could win big! http://www.GETREALPhotoContest.com?ocid=TXT_TAGHMloc=us ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi problem: offline peers still in request EID/EID_DIRECT field?
hi all! I have the following situation: 1 2 ¦¦ ¦¦ 3--4 ¦¦ ¦¦ 5--6 where 1 ... 6 are nodes and every direct neighbor is specified as a dundi peer (in *). When I start a dundi request, every queried node is mentioned in the dpdiscover. For example 1 sends a discover to 2 and 3, so 2 sees in the EID or EID_DIRECT field that a discover has also been sent to 3. So much for that. Is this field also filled with the neighbour peers even if they are unreachable/offline? 1 ---x 2 ¦¦ ¦¦ 3x-4 ¦¦ ¦¦ 5--6 My problem: When links break (e.g. 1-2 and 3-4) I have the problem that 6 doesn't forward the query (received from 5) to 4, because 4 is mentioned in the EID or EID_DIRECT field even though it is not possible that this peer could have been reached. Is this a problem of the protocol or can I fix this by setting a special option in *? Thanks for helping. Best regards Andre___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] features.conf / DTMF / automon hell
I have been trying for a very long time to get asterisk to detect and utilize dtmf tones from my sip clients within my dial scripts. I have set automon=#9 in my features.conf, I have Dial(,gWw) in my dial scripts. I have Set(DYNAMIC_FEATURES=automon) as the first script in my extension. I can see the dtmf tones on the wire as SIP INFO packets. Using the Read() app I have verified that * is in fact understanding the dtmf info packets from the sip phone (the read app works). I have verified that the Monitor() app is present and works. I just can not get * to do anything from my features.conf file. I have also done include= featuremap. I have been all over the web, posted multiple times on the irc channels. Could someone please help here. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Warnnings
You're including a context in your dialplan that doesn't exist. Given that it has been prefixed with AEL, I'd check extensions.ael for the Asterisk Extension Language sample file. I bet it does some including. Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Umar Draz Sent: Friday, June 29, 2007 10:19 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.4 Warnnings Dear Users ! I have recently installed asterisk 1.4 i got a warning message whenever i use reload or extensions reload. [Jun 29 19:22:11] WARNING[4539]: pbx.c:6236 ast_context_verify_includes: Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls' [Jun 29 19:22:11] WARNING[4539]: pbx.c:6236 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-canonical' [Jun 29 19:22:11] WARNING[4539]: pbx.c:6236 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-customers' [Jun 29 19:22:11] WARNING[4539]: pbx.c:6236 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-via-pstn' please help what is this and how i can fix it? Regards, Umar Draz _ Picture this - share your photos and you could win big! http://www.GETREALPhotoContest.com?ocid=TXT_TAGHMloc=us ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel Compilation
Hi Steve; I did what I told me below, and look like going fine but I do not know how can I know that zaptel compilation was implemented successfully specially I do not have a message in the end indicate this, please find below what the make and make install commands (for zaptel compilation) was ended by (please let me know if that is normal and the compilation was successfully done): This for make: make[2]: Leaving directory `/usr/src/asterisk-1.4/zaptel-1.4/xpp/utils' gcc -c -fPIC -DSTANDALONE_ZAPATA -DBUILDING_TONEZONE -DHOTPLUG_FIRMWARE -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DSTANDALONE_ZAPATA -DBUILDING_TONEZONE -DHOTPLUG_FIRMWARE -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c gcc -c -fPIC -DSTANDALONE_ZAPATA -DBUILDING_TONEZONE -DHOTPLUG_FIRMWARE -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DSTANDALONE_ZAPATA -DBUILDING_TONEZONE -DHOTPLUG_FIRMWARE -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c gcc -shared -Wl,-soname,libtonezone.so.1.0 -o libtonezone.so zonedata.lo tonezone.lo -lm make[1]: Leaving directory `/usr/src/asterisk-1.4/zaptel-1.4' This for make install: *** *** WARNING: *** If you had custom settings in /etc/modprobe.d/zaptel, *** they have been moved to /etc/modprobe.d/zaptel.bak. *** *** In the future, do not edit /etc/modprobe.d/zaptel, but *** instead put your changes in another file *** in the same directory so that they will not *** be overwritten by future Zaptel updates. *** Regards Bilal bilal ghayyad wrote: Hi List; Maybe I have to resummarize my problem with Zaptel Compilation: I am getting the error while I am compilaing Zaptel when I ran the command make linux26, although I did the: software symbolic link, ./configure, and I checked my kernel is 2.6.20-1.2320.fc5 which typical for the output of cat cat/proc/version. What should I do more? Please find below all the results of the above steps, and I hope if any one can help me and advise me where is my mistake or what else I have to do? 1) I have a soft symbolic link and we can see it as following: [EMAIL PROTECTED] /]# cd lib/modules/2.6.20-1.2320.fc5 [EMAIL PROTECTED] 2.6.20-1.2320.fc5]# ls -l total 1328 lrwxrwxrwx 1 root root 47 Jun 24 23:47 build - ../../../usr/src/kernels/2.6 .20-1.2320.fc5-i686 drwxr-xr-x 2 root root 4096 Jun 13 02:28 extra drwxr-xr-x 9 root root 4096 Jun 24 23:47 kernel -rw-r--r-- 1 root root 280802 Jun 24 23:47 modules.alias -rw-r--r-- 1 root root 69 Jun 24 23:47 modules.ccwmap -rw-r--r-- 1 root root 277363 Jun 24 23:47 modules.dep -rw-r--r-- 1 root root813 Jun 24 23:47 modules.ieee1394map -rw-r--r-- 1 root root206 Jun 24 23:47 modules.inputmap -rw-r--r-- 1 root root 12236 Jun 24 23:47 modules.isapnpmap -rw-r--r-- 1 root root 216224 Jun 24 23:47 modules.pcimap -rw-r--r-- 1 root root967 Jun 24 23:47 modules.seriomap -rw-r--r-- 1 root root 121611 Jun 24 23:47 modules.symbols -rw-r--r-- 1 root root 332324 Jun 24 23:47 modules.usbmap lrwxrwxrwx 1 root root 5 Jun 24 23:47 source - build drwxr-xr-x 2 root root 4096 Jun 13 02:28 updates 2) I did ./configure successfully as following: [EMAIL PROTECTED] /]# cd usr/src/asterisk-1.4.4/zaptel-1.4.2.1 [EMAIL PROTECTED] zaptel-1.4.2.1]# ./configure checking for gcc... gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... no checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether gcc accepts -g... yes checking for gcc option to accept ISO C89... none needed checking how to run the C preprocessor... gcc -E checking for a BSD-compatible install... /usr/bin/install -c checking whether ln -s works... yes checking for GNU make... make checking for grep... /bin/grep checking for sh... /bin/sh checking for ln... /bin/ln checking for wget... /usr/bin/wget checking for grep that handles long lines and -e... (cached) /bin/grep checking for egrep... /bin/grep -E checking for ANSI C header files... yes checking for sys/types.h... yes checking for sys/stat.h... yes checking for stdlib.h... yes checking for string.h... yes checking for memory.h... yes checking for strings.h... yes checking for inttypes.h... yes checking for stdint.h... yes checking for unistd.h... yes checking for initscr in -lcurses... yes checking curses.h usability... yes checking curses.h presence... yes checking for curses.h... yes checking for initscr in -lncurses... yes checking for curses.h... (cached) yes checking for newtBell in -lnewt... yes checking newt.h usability... yes checking newt.h
Re: [asterisk-users] Linking Asterisk with another SIP PBX (or SIP Softswitch)
Yes, you can only send calls to peers, not receive them, so no context= needed. Moj bilal ghayyad wrote: Hi Noah; The reason that I am asking wether I need to determine the context is what I read in the documentation (about configuring outbound IAX connections), it did not mention the context at all, please read the below paragraph (I copy it from the documentation and paste it): Configuring Outbound IAX Connections: While an IAX user receives inbound calls; an IAX peer is used to place outbound calls. This section will set up iax.conf and extensions.conf so that you can place calls. iax.conf Configuration: The following entry in iax.conf can be used to place a call on the FWD network: [iaxfwd] type=peer host=iax2.fwdnet.net username=fwd-account-number secret=fwd-account-password qualify=yes disallow=all allow=ulaw allow=gsm allow=ilbc allow=g726 As you see that no context written in the iaxfwd peer configuration, so how it will work? Is it because the type is peer? Regards Bilal If I need to do a trunk between Asterisk and another SIP softswitch (so Asterisk will send a SIP calls to that softswitch), then I have to configure this on the sip.conf file Yes. And is it the same when I configure iax trunk? Not exactly the same, but very close. Here's a page on how to connect Asterisk to Cisco Call Manager using SIP: http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration Should I determine the context in this case for this SIP trunk? Yes. - Noah Need a vacation? Get great deals to amazing places on Yahoo! Travel. http://travel.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + hinting presence + macro
Eric ManxPower Wieling wrote:
Rob Schall wrote:
Eric ManxPower Wieling wrote:
Rob Schall wrote:
I currently have about 50 polycom 501 phones on my asterisk setup.
The dialplan is set to work with mysql (realtime), and all of the
extensions for the phones route through the same macro (stdexten).
This all works fine until I tried to set up notify status.
On voip-info, they say do something like...
,hint,SIP/
,1,Dial(SIP/)
blah blah blah
This functionality works fine. But what if you have a macro
s,hint,SIP/${ARG1}
s,1,Dial(SIP/${ARG1}
this adds a s hint which obviously doesn't work, instead of a hint
for as it should.
Yes. Put in the correct hint. There is no reason that
,hint,SIP/ would not work in a macro.
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So, if I understand you correctly, my macro would look something vaguely
like...
[macro-stdexten]
${ARG1},hint,SIP/${ARG1}
s,1,Dial(${ARG1})?
This will work? My understand was that by going into a macro, you were
going to be using the s extension. I'm not sure how that hint would
get called if its not inside the s extension.
I have no idea, but as I understand it, Hints are separate from extensions.
I guess you could do something like:
[macro-stdexten]
exten = s,1,Goto(${MACRO_EXTEN},1)
exten = _,hint,SIP/${ARG1}
exten = _,1,Dial(${ARG1})
I do this sort of thing in many of my macros that Dial somewhere. I
seem to remember something about hints not working for pattern matching.
or working weirdly.
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Hints do not have to be part of the steps in a dialplan, and if you want
devices to be listeners, they shouldn't be. Also you had stated you
where using realtime, and the hint priority doesnt work in realtime, I
submitted a bug about this and it was rejected as won't fix, but was
explained as, not going to be fixed anytime soon.
At an rate make a hint context that has devices mapped to extensions,
[hint-context]
,hint,Sip/
4445,hint,Sip/4445
and the like, then add that context in sip.conf for all devices both
monitored and listeners,
subscribecontext=hint-context
Then ast will send update notifications about these devices to watchers
properly.
Hope this helps,
Anthony
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[asterisk-users] Voice Mail not Receive
hi,
i am using Asterisk 1.4. and unable to get Voice Mail below is my config
extensions.conf
exten = 50,1,NoOp(Failover)
exten = 50,2,MixMonitor(${CALLERID(num)}_${TIMESTAMP}.gsm,b)
exten = 50,3,Dial(SIP/101,18)
exten = 50,4,Goto(ss-${DIALSTATUS},1)
exten = ss-NOANSWER,1,StopMixMonitor()
exten = ss-NOANSWER,n,VoiceMail([EMAIL PROTECTED])
voicemail.conf
[salesvoice]
777 = 1212, sales, [EMAIL PROTECTED]
with same setting i m getting voice mail when i use Asterisk-1.2 but
when i use Asterisk-1.4 i m not able to get a voice mail with these
setting.
Please help me regarding this issue.
thanks
Muhammad Asif
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[asterisk-users] v1.4.x ready yet?
Hi All, Eagerly waiting for v1.4.x to mature a bit before getting serious about it. Is it ready for production yet? If that's too general, where is it in terms of stability compared to where 1.2.x is now. Anyone running it successfully in production environment and if so what sort of config do you have? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold 1.2
On 6/29/07, Ed Nuñez [EMAIL PROTECTED] wrote: What is a good solution for playing music on hold on the 1.2 branch. I do not want to use mpg123 because last time I used it in a production server it caused many problems. The MPG123 process was taking about 60% of my Xeon CPU. For minimum system resource usage, convert your MP3 files into the same codec as most of your users' phones are set to (alaw/ulaw perhaps), and read the musiconhold.conf file for details of how to use them. Cheers, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Music on hold 1.2
What is a good solution for playing music on hold on the 1.2 branch. I do not want to use mpg123 because last time I used it in a production server it caused many problems. The MPG123 process was taking about 60% of my Xeon CPU. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .call file
I'm going to top post in this situation.
Kevin - Commands that operate on the channel variables won't help if we
are using a call file. We will have a new channel.
This syntax works with asterisk 1.2.x for me:
Application: AGI
Data: say_it.php|call_status_message
I have done other things where a bunch of parameters are stored in
postgres or mysql and the only parameter I pass via the call file is the
record key. The php script receives the key as a parameter and gets
everything else from the db. Something like this:
Application: AGI
Data: inform.php|68456943
Kevin Smith wrote:
Nitesh Divecha wrote:
Hello All,
Is there any way to pass additional parameters while calling AGI from
*.call file?
Channel: Local/[EMAIL PROTECTED]
MaxRetries: 0
RetryTime: 15
WaitTime: 15
Application: AGI
Data: recordvoice.php
Something like Data: recordvoice.php?id=3453name=asterisk
Cheers,
Nitesh
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I'm not 100% sure if you can pass it directly, but you can use the set
option in the call file to set local variables within Asterisk and then
pass them to the AGI script. So for your example it would be.
Set: name=asterisk
This will set the variable ${name} in asterisk and depending how your
script was created you should be able to grab the variable to use within
the script. If you are using say the PHP AGI you can use something like
the following:
$var = $agi-get_variable(name);
This will create an array with $var['data'] holding 'asterisk';
Now one more thing I am not sure of is for multiple variables (haven't
tried it yet ;D ). You may have to do it one of two ways.
Set: name=asterisk, id=3453
or
Set: name=asterisk
Set: id=3453
and if those don't work, just format it so you can filter it out with PHP.
Hopefully this will help.
Kevin
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Re: [asterisk-users] Asterisk + hinting presence + macro
Anthony Francis wrote:
Eric ManxPower Wieling wrote:
Rob Schall wrote:
Eric ManxPower Wieling wrote:
Rob Schall wrote:
I currently have about 50 polycom 501 phones on my asterisk setup.
The dialplan is set to work with mysql (realtime), and all of the
extensions for the phones route through the same macro (stdexten).
This all works fine until I tried to set up notify status.
On voip-info, they say do something like...
,hint,SIP/
,1,Dial(SIP/)
blah blah blah
This functionality works fine. But what if you have a macro
s,hint,SIP/${ARG1}
s,1,Dial(SIP/${ARG1}
this adds a s hint which obviously doesn't work, instead of a hint
for as it should.
Yes. Put in the correct hint. There is no reason that
,hint,SIP/ would not work in a macro.
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So, if I understand you correctly, my macro would look something vaguely
like...
[macro-stdexten]
${ARG1},hint,SIP/${ARG1}
s,1,Dial(${ARG1})?
This will work? My understand was that by going into a macro, you were
going to be using the s extension. I'm not sure how that hint would
get called if its not inside the s extension.
I have no idea, but as I understand it, Hints are separate from extensions.
I guess you could do something like:
[macro-stdexten]
exten = s,1,Goto(${MACRO_EXTEN},1)
exten = _,hint,SIP/${ARG1}
exten = _,1,Dial(${ARG1})
I do this sort of thing in many of my macros that Dial somewhere. I
seem to remember something about hints not working for pattern matching.
or working weirdly.
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Hints do not have to be part of the steps in a dialplan, and if you want
devices to be listeners, they shouldn't be. Also you had stated you
where using realtime, and the hint priority doesnt work in realtime, I
submitted a bug about this and it was rejected as won't fix, but was
explained as, not going to be fixed anytime soon.
At an rate make a hint context that has devices mapped to extensions,
[hint-context]
,hint,Sip/
4445,hint,Sip/4445
and the like, then add that context in sip.conf for all devices both
monitored and listeners,
subscribecontext=hint-context
Then ast will send update notifications about these devices to watchers
properly.
Hope this helps,
Anthony
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Anthony,
I didn't realize you could do hints like that. This will be very
interesting if I can make it work
In extensions.conf, I added
[hint-context]
,hint,SIP/
5053,hint,SIP/5053
I think added an include to my main context (internal). I'm assuming
these need to be included.
In sip.conf, I added:
subscribecontext = hint-context
notifyringing = yes
and in the phone directories, I added: bw#/bw to the corresponding
entries.
However, the phones show as offline and using show hints from the
cli also says no hints.
Any ideas?
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Re: [asterisk-users] v1.4.x ready yet?
While I have not jumped all my systems to 1.4, there were some that I have moved to 1.4 and I have found it to be as stable as 1.2 was on those machines.One of the systems is a 10 user office with Sangoma cards and another is a 70+ user pure voip system. In both cases I have warning messages about my dialplan usage of realtime and the fact that it will be depreciated in the next release, but everything works as it should and the upgrades.txt guided me through the changes to my dialplan. Hope that helps. On 6/29/07, shadowym [EMAIL PROTECTED] wrote: Hi All, Eagerly waiting for v1.4.x to mature a bit before getting serious about it. Is it ready for production yet? If that's too general, where is it in terms of stability compared to where 1.2.x is now. Anyone running it successfully in production environment and if so what sort of config do you have? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kore dump
For anyone interested on the crashes I was experiencing when using ChanSpy from SIP extension to SIP extensions with the group option. For the last couple of days, Ive been monitoring from Zap extensions to SIP extensions, and the system has not crashed once. The problem only happens when I spy from SIP. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vadim Berezniker Sent: Tuesday, June 26, 2007 2:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] kore dump use the safe_asterisk script it will restart asterisk if it crashes and it enables core dumps (your core size limit is probably set to 0 when you start asterisk). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez Sent: Tuesday, June 26, 2007 2:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: [asterisk-users] kore dump I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server. My PBX has experienced several core dumps the last couple of days and I am not sure if this is whats causing it, but it always seems to happen when a particular extension on a grandstream phone uses ChanSpy SIP group. I have not been able to locate where the core dump file is being saved. I cant find it in my TMP directory. I would also like to know if Asterisk can be setup to automatically re start if there is a core dump. I was thinking of setting up a cron job to launch Asterisk every minute. If its running, no harm done, and if it crashes, the cron job will make sure that its started every 60 seconds. Any suggestions? Thank you Ed Nuñez ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Music on hold - 1.4.5
Hi, Please bear with me if I'm asking stupid questions... I'm new to Asterisk, newish to Linux, etc... I've got MoH working nicely with my new Asterisk setup using the files option; except that it always plays from the start of a (random) music file when you first put someone on hold. Take them off hold put them back, and sometimes (not always!) it will start playing a new file from the beginning If I park a call, from the point of pressing the TRNF button the caller gets music; but, when the call parks, the music starts a new file! What I'd like to do is have the music streaming constantly, so the on hold caller always gets music at the current position; even if that's in the middle or near the end of a file. The musiconhold.conf file mentions a couple of streaming options; but (rightly) doesn't go into particular detail. So, what's my best strategy? For info: - Asterisk is running on a P3 1GHz server (it's only a tiny experimental PBX setup though) - v1.4.5, compiled by myself (thanks to voip-info.org a couple of other sites) - Server is Ubuntu Fiesty Fawn, clean install (especially for Asterisk) - VoIP (SIP) only - All music files are in uLaw format, and the SIP phones are forced to use uLaw encoding. Cheers! Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.9.12/878 - Release Date: 28/06/2007 17:57 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE420 PCI Express Card
Hi, Does anybody have any feedback on this new card from Digium? It was announced a couple of weeks ago but now Digium said they ran out and is no longer available for purchase via their web site. I find this kind off odd. If they are out why don't they just says its on backorder or something, rather than pulling it off completely from the store. I am kind of hesistant at this point in recomending the card to our customers. Thanks, -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX over T1
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andres Paglayan Sent: Friday, June 29, 2007 12:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX over T1 On Jun 22, 2007, at 3:43 PM, Joe acquisto wrote: I have an existing Hylafax system using a mainpine 4 port board, 4 POTS lines. Have a recently installed Asterisk system, with a dedicated T1 line. (Well, it's really a fonality system). What would I need to do, or where is the reading material, for what I need to do, to convert the Hylafax server to use the T1 line? Reliably. Preferably to use DID's as well. The current FAX works fine, but there is some desire to get rid of the analog lines. Could one add some sort of device in the Asterisk server, to act as FAX extensions, keeping the mainpine on the hylafax? Like a TDM400p with FSX modules? I'm just saying, ya know? I suppose I have to ask fonality, since it's their box? joe a. did you fix this yet? I had the same problem, and worked it out, contact me off list if you want the how to (or at least one of the how tos) Andres Paglayan --Harmony is more important than being right Bapak Why not post the how-tos to the list for anyone down the road that is searching for this same issue? right, mostly because if it's fixed, I don't like writing with no purpose, cause we are using a fonality box, and they don't support any kind of faxing whatsoever, (although their sales people will tell you otherwise) nor will let you run any custom mod in the same box without breaching the support agreement, (although their sales people will tell you otherwise) I decided to run an external fax-server, the solution is very straight forward, create virtual extension for each incoming did number that is dedicated to fax create sub-menus for each virtual fax number, edit those to set the caller ID to match either their own or the DID, (you'll use the callerID to route the fax) and then transfer to the fax extension, the fax extension goes to a modem-fax plugged to your hylafax-server we use multitech modems because they work great, then edit FaxDispatch on hylafax to route faxes to email according to caller ID Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem getting a Perl script to run
I have Apache2 set up and running on a system I only use for testing. In trying to access a script that is an html and only points to a Perl script. When it reaches the Perl script I get this message: You have chosen to open filename.pl which is a perl script from --- What should I do with this file? Then I get options which are open with edit other I am running Firefox 2.0 to access the apache script. I am fairly new to Apache and my searches seem not to find an answer. I am hoping someone here can point me to what is wrong. Thanks for any help you can offer. Bob R ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linking Asterisk with another SIP PBX
So if the it is only outgoing then no need for context but if it is incoming or incmoing outgoing then I need context. Correct? Regards Bilal Yes, you can only send calls to peers, not receive them, so no context= needed. Moj Need a vacation? Get great deals to amazing places on Yahoo! Travel. http://travel.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk call unique id in dialplan
many thanks!
bye
On 6/29/07, Alexander Lopez [EMAIL PROTECTED] wrote:
In the top directory of your asterisk source in the doc dir there is a
file that explains channel variables.
From that file:
${UNIQUEID} * Current call unique identifier
BEWARE the UNIQUEID can be repeated do not use this as a primary index
on your databse.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of nik600
Sent: Friday, June 29, 2007 6:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] asterisk call unique id in dialplan
Hi
how can i retrieve the call unique id of asterisk in the dialplan?
I have enabled the cdr logging on a postgres database.
In the table cdr each record has a field that assumes an unique id
(for example: 1141628669.51)
Can i retrieve this from the dialplan?
For example:
exten = 203,1,Answer
exten = 203,2,Set(CALLERID(name)=UNIQUE_ID - ${var_name_unique_id})
exten = 203,3,Dial(SIP/203)
Can i do something similar that?
How can i retrieve the unique_id generated?
thanks.
--
/*/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser
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--
/*/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser
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Re: [asterisk-users] Asterisk + hinting presence + macro
Rob Schall wrote:
Anthony Francis wrote:
Eric ManxPower Wieling wrote:
Rob Schall wrote:
Eric ManxPower Wieling wrote:
Rob Schall wrote:
I currently have about 50 polycom 501 phones on my asterisk setup.
The dialplan is set to work with mysql (realtime), and all of the
extensions for the phones route through the same macro (stdexten).
This all works fine until I tried to set up notify status.
On voip-info, they say do something like...
,hint,SIP/
,1,Dial(SIP/)
blah blah blah
This functionality works fine. But what if you have a macro
s,hint,SIP/${ARG1}
s,1,Dial(SIP/${ARG1}
this adds a s hint which obviously doesn't work, instead of a hint
for as it should.
Yes. Put in the correct hint. There is no reason that
,hint,SIP/ would not work in a macro.
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So, if I understand you correctly, my macro would look something vaguely
like...
[macro-stdexten]
${ARG1},hint,SIP/${ARG1}
s,1,Dial(${ARG1})?
This will work? My understand was that by going into a macro, you were
going to be using the s extension. I'm not sure how that hint would
get called if its not inside the s extension.
I have no idea, but as I understand it, Hints are separate from extensions.
I guess you could do something like:
[macro-stdexten]
exten = s,1,Goto(${MACRO_EXTEN},1)
exten = _,hint,SIP/${ARG1}
exten = _,1,Dial(${ARG1})
I do this sort of thing in many of my macros that Dial somewhere. I
seem to remember something about hints not working for pattern matching.
or working weirdly.
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Hints do not have to be part of the steps in a dialplan, and if you want
devices to be listeners, they shouldn't be. Also you had stated you
where using realtime, and the hint priority doesnt work in realtime, I
submitted a bug about this and it was rejected as won't fix, but was
explained as, not going to be fixed anytime soon.
At an rate make a hint context that has devices mapped to extensions,
[hint-context]
,hint,Sip/
4445,hint,Sip/4445
and the like, then add that context in sip.conf for all devices both
monitored and listeners,
subscribecontext=hint-context
Then ast will send update notifications about these devices to watchers
properly.
Hope this helps,
Anthony
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Anthony,
I didn't realize you could do hints like that. This will be very
interesting if I can make it work
In extensions.conf, I added
[hint-context]
,hint,SIP/
5053,hint,SIP/5053
I think added an include to my main context (internal). I'm assuming
these need to be included.
In sip.conf, I added:
subscribecontext = hint-context
notifyringing = yes
and in the phone directories, I added: bw#/bw to the corresponding
entries.
However, the phones show as offline and using show hints from the
cli also says no hints.
Any ideas?
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I don't know about the phone directories you are using, but as long as
those are the actual sip id's of the devices it should work, oh and to
get it all to work after putting it in the dialplan you have to issue a
reload command in the CLI.
Here is an example (real-world) where I am using it and showing sip
pressence on a Aastra 55i.
[rockynet-1000-hint]
exten = 865,hint,SIP/7CEC9925-865SIP/27E68FA6-865
exten = 3036292865,hint,SIP/5D03CB01-865
exten = 864,hint,SIP/B7AA2A4A-864
exten = 858,hint,SIP/27623324-858
exten = 861,hint,SIP/atabryan
exten = 3036292861,hint,SIP/E30CD3F6-861
exten = 857,hint,SIP/F7D4BFD1-857
exten = 868,hint,SIP/B956285D-868
exten = 863,hint,SIP/625AB6D7-863
exten = 863,hint,SIP/625AB6D7-863
exten = 854,hint,SIP/CE74050E-854
exten = 866,hint,SIP/7CDF7A01-866
in sip.conf
subribecontext=rockynet-1000-hint
The device that is subscribing has to have that entry as well.
This example works in production in a
Re: [asterisk-users] Problem getting a Perl script to run
Sounds like your filename.pl script should be in a cgi-bin directory rather than in a document directory? How exactly are you doing this from asterisk? Is this for a microbrowser in a desk phone? Moj Robert A. Rawlinson wrote: I have Apache2 set up and running on a system I only use for testing. In trying to access a script that is an html and only points to a Perl script. When it reaches the Perl script I get this message: You have chosen to open filename.pl which is a perl script from --- What should I do with this file? Then I get options which are open with edit other I am running Firefox 2.0 to access the apache script. I am fairly new to Apache and my searches seem not to find an answer. I am hoping someone here can point me to what is wrong. Thanks for any help you can offer. Bob R ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem getting a Perl script to run
Opps! Sorry wrong list Robert A. Rawlinson wrote: I have Apache2 set up and running on a system I only use for testing. In trying to access a script that is an html and only points to a Perl script. When it reaches the Perl script I get this message: You have chosen to open filename.pl which is a perl script from --- What should I do with this file? Then I get options which are open with edit other I am running Firefox 2.0 to access the apache script. I am fairly new to Apache and my searches seem not to find an answer. I am hoping someone here can point me to what is wrong. Thanks for any help you can offer. Bob R ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.20 and 1.4.6 released
The Asterisk development team is proud to announce the releases of versions 1.2.20 and 1.4.6! These releases are regular maintenance releases. They have been made just a couple of weeks after the previous set of releases because the development team has been working especially hard on fixing bugs lately. There has been a large volume of issues fixed in just two weeks. We would also like to continue to encourage the community to upgrade to the 1.4 series. There have been almost 100 changes to the 1.4 tree since the last release. Keep in mind that we are still planning to move the 1.2 series of Asterisk into security maintenance only beginning August 1st. Thank you for your support! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] awful list delays: 4 days!
Hello list, I am getting the list with days of delay, take for example this message: Received: from unknown (HELO lists.digium.com) (216.207.245.17) by mxavas16.fe.aruba.it with SMTP; 29 Jun 2007 13:38:37 - Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) by lists.digium.com with esmtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I2zQW-0004ty-N8; Mon, 25 Jun 2007 20:01:04 -0500 Received: from exprod8mx64.postini.com ([64.18.3.164] helo=psmtp.com) by lists.digium.com with smtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I2zQL-0004tU-PU for asterisk-users@lists.digium.com; Mon, 25 Jun 2007 20:00:54 -0500 As you can see, the message was posted on June 25th and was sent to my email on June 29th! am I the only one who is getting such an awful message turn-around time? l. -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] v1.4.x ready yet?
I jumped into asterisk 1.4 and its pretty stable .. i never got a core dump but it did halt while reloading a few times . I am back on asterisk 1.2 now but i think asterisk 1.4 is stable . On 29/06/07, Bruce Reeves [EMAIL PROTECTED] wrote: While I have not jumped all my systems to 1.4, there were some that I have moved to 1.4 and I have found it to be as stable as 1.2 was on those machines.One of the systems is a 10 user office with Sangoma cards and another is a 70+ user pure voip system. In both cases I have warning messages about my dialplan usage of realtime and the fact that it will be depreciated in the next release, but everything works as it should and the upgrades.txt guided me through the changes to my dialplan. Hope that helps. On 6/29/07, shadowym [EMAIL PROTECTED] wrote: Hi All, Eagerly waiting for v1.4.x to mature a bit before getting serious about it. Is it ready for production yet? If that's too general, where is it in terms of stability compared to where 1.2.x is now. Anyone running it successfully in production environment and if so what sort of config do you have? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE420 PCI Express Card
Andres wrote: Does anybody have any feedback on this new card from Digium? It was announced a couple of weeks ago but now Digium said they ran out and is no longer available for purchase via their web site. I find this kind off odd. If they are out why don't they just says its on backorder or something, rather than pulling it off completely from the store. I expect that there has been an extremely high demand for these cards by Digium distributors, and that is why they are not available from Digium directly at the moment. Have you tried contacting any official distributors or resellers? -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] awful list delays: 4 days!
On Jun 29, 2007, at 12:50 PM, Lenz wrote: Hello list, I am getting the list with days of delay, take for example this message: As you can see, the message was posted on June 25th and was sent to my email on June 29th! am I the only one who is getting such an awful message turn-around time? l. I'll let you know next week, ;^) -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Andres Paglayan --Harmony is more important than being right Bapak ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE420 PCI Express Card
Hi Russel, I called on official distributor in the Miami area (Commlogik) and they did not have it at the moment. I also tried several on-line resellers like Voipsupply, Telephony Depot, Voxilla, and they do not have it either. I was preparing a quote for a customer and was planning on including this card but it seems like it is nowhere to be found. Thanks, Andres Russell Bryant wrote: Andres wrote: Does anybody have any feedback on this new card from Digium? It was announced a couple of weeks ago but now Digium said they ran out and is no longer available for purchase via their web site. I find this kind off odd. If they are out why don't they just says its on backorder or something, rather than pulling it off completely from the store. I expect that there has been an extremely high demand for these cards by Digium distributors, and that is why they are not available from Digium directly at the moment. Have you tried contacting any official distributors or resellers? -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] awful list delays: 4 days!
Andres Paglayan wrote: On Jun 29, 2007, at 12:50 PM, Lenz wrote: Hello list, I am getting the list with days of delay, take for example this message: As you can see, the message was posted on June 25th and was sent to my email on June 29th! am I the only one who is getting such an awful message turn-around time? l. I'll let you know next week, ;^) -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Andres Paglayan --Harmony is more important than being right Bapak ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ROFL, yeah its you. I see posts within a few hours. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] awful list delays: 4 days!
Is it taking a while for _your_ messages to post to the list, or do you mean messages from the mailing list software take days to get to you? Moj Lenz wrote: Hello list, I am getting the list with days of delay, take for example this message: Received: from unknown (HELO lists.digium.com) (216.207.245.17) by mxavas16.fe.aruba.it with SMTP; 29 Jun 2007 13:38:37 - Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) by lists.digium.com with esmtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I2zQW-0004ty-N8; Mon, 25 Jun 2007 20:01:04 -0500 Received: from exprod8mx64.postini.com ([64.18.3.164] helo=psmtp.com) by lists.digium.com with smtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I2zQL-0004tU-PU for asterisk-users@lists.digium.com; Mon, 25 Jun 2007 20:00:54 -0500 As you can see, the message was posted on June 25th and was sent to my email on June 29th! am I the only one who is getting such an awful message turn-around time? l. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] v1.4.x ready yet?
shadowym wrote: Eagerly waiting for v1.4.x to mature a bit before getting serious about it. Is it ready for production yet? If that's too general, where is it in terms of stability compared to where 1.2.x is now. Anyone running it successfully in production environment and if so what sort of config do you have? The release of 1.4.6 today had almost 100 changes to the 1.4 tree since the last release. That says to me that the answer is no. I'll wait a week or two and then check the bug tracker for outstanding issues (as well as the changelog) for 1.4 and then decide if I think it is worth trying to upgrade or not. I also hang out on #asterisk IRC channel to see what sorts of problems people have with 1.4. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: fail to load modules
Hi, Does Asterisk_addons_1.4.2 cant be use in the new Asterisk release or no one really want to share their experiences? Since this project is belonging to everyone within this list, why still no one really want to share the experiences and to growth the Asterisk to the next level by keeping their secret in behind. As see, Asterisk 1.4 has so many feature improvements, and it's functionality are almost there as compare with the world leader like Alcatel-Lucent, Avaya and Nortel, let us share the basic knowledge so that we can commit to this project. Sad with those who know but don't share. _ From: clive.chan(Alpha Trilogies Networks) [mailto:[EMAIL PROTECTED] Sent: Thursday, June 28, 2007 5:21 PM To: 'asterisk-users@lists.digium.com' Subject: fail to load modules Hi all, I am a bit out with the Asterisk 1.4.4, after I complied and installed the Asterisk and I got such error messages [Jun 28 16:56:19] WARNING[28625] res_smdi.c: No SMDI interfaces are available to listen on, not starting SDMI listener. [Jun 28 16:56:19] WARNING[28625] loader.c: Error loading module 'chan_ooh323.so': /usr/lib/asterisk/modules/chan_ooh323.so: undefined symbol: ast_rtp_bridge [Jun 28 16:56:19] WARNING[28625] loader.c: Error loading module 'res_config_mysql.so': /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: option_verbose I got nothing error during installation of asterisk-addons-1.4.2 after I had change the Make file on the chan_ooh323.so.1.0.1. Tried; I tried to define noload to the chan_00h323.so and res_config_mysql.so, my asterisk start but give me others problems as bellowing... [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'cdr_addon_mysql.so' did not register itself during load [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'cdr_addon_mysql.so' could not be loaded. [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_addon_sql_mysql.so' did not register itself during load [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_addon_sql_mysql.so' could not be loaded. [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'format_mp3.so' did not register itself during load [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'format_mp3.so' could not be loaded. [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_saycountpl.so' did not register itself during load [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_saycountpl.so' could not be loaded. Can some one shares experience ?? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: fail to load modules
clive.chan(Alpha Trilogies Networks) wrote: I tried to define noload to the chan_00h323.so and res_config_mysql.so, my asterisk start but give me others problems as bellowing... [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'cdr_addon_mysql.so' did not register itself during load [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'cdr_addon_mysql.so' could not be loaded. [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_addon_sql_mysql.so' did not register itself during load [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_addon_sql_mysql.so' could not be loaded. [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'format_mp3.so' did not register itself during load [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'format_mp3.so' could not be loaded. [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_saycountpl.so' did not register itself during load [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_saycountpl.so' could not be loaded. It sounds like you have a version mismatch between Asterisk and Asterisk-addons. The quick way to solve this would be to clean it up a little bit and start over. # rm /usr/lib/asterisk/modules/* # rm -rf /usr/include/asterisk # cd src/asterisk-1.4.6 # ./configure make make install # cd src/asterisk-addons/1.4.2 # ./configure make make install Then, everything should be happy. -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kore dump
Ed Nuñez wrote: For anyone interested on the crashes I was experiencing when using ChanSpy from SIP extension to SIP extensions with the group option. For the last couple of days, I’ve been monitoring from Zap extensions to SIP extensions, and the system has not crashed once. The problem only happens when I spy from SIP. First, just to be safe, give 1.4.6 a try. If you still have a problem, please report it to bugs.digium.com We would be happy to help figure it out and get it fixed. To get the backtrace: 1) Run make menuselect, go to Compiler Flags, select DONT_OPTIMIZE. 2) # make clean ; make ; make install 3) Run asterisk with the -g argument 4) gdb /usr/sbin/asterisk core.12345 4.i) (gdb) bt 4.ii) (gdb) bt full -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Problems with zap following 1.4.6 install
Hi, Just upgrading to 1.4.6 from 1.2. SIP channels work OK, but not zap. I have a TDM400 w/ an FXO 2 FXS. I built libpri 1.4.0 first then zaptel 1.4.3. Menuselect had a * beside chan_zap and I loaded the wcusb wctdm before building asterisk. In the CLI zap show channels returns no such command. chan_zap.so exists in /usr/lib/asterisk/modules. What could be the problem how to fix? Thanks, Hugh zapata.conf = [channels] ; language=en context=outgoing-PBX signalling=fxo_ks threewaycalling=yes transfer=yes mailbox = [EMAIL PROTECTED],[EMAIL PROTECTED],[EMAIL PROTECTED] group=1 callgroup=1 pickupgroup=1,2 channel=1 ; language=en context=ils-outgoing-PBX signalling=fxo_ks threewaycalling=yes transfer=yes mailbox = [EMAIL PROTECTED] group=1 callgroup=2 pickupgroup=1,2 channel=2 ; ;** ; Incoming channels ;** usedistinctiveringdetection=yes dring1=384,327,0 dring1context=ils-incoming ; dring2=0,0,0 dring2context=smith-incoming ; language=en context=incoming-zap signalling=fxs_ks usecallerid=yes echocancel=yes echocancelwhenbridged=yes transfer=yes ;rxgain=5% group=2 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Problems with zap following 1.4.6 install
hugolivude wrote: Just upgrading to 1.4.6 from 1.2. SIP channels work OK, but not zap. I have a TDM400 w/ an FXO 2 FXS. I built libpri 1.4.0 first then zaptel 1.4.3. Menuselect had a * beside chan_zap and I loaded the wcusb wctdm before building asterisk. In the CLI zap show channels returns no such command. chan_zap.so exists in /usr/lib/asterisk/modules. What could be the problem how to fix? What output do you get if you run module unload chan_zap.so and then module load chan_zap.so ? -- Russell Bryant Software Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
