from:"broadbandvoice"


When I started Asterisk I get this error but it is working fine and should I be concerned. Error below:

[EMAIL PROTECTED] asterisk]# /etc/rc.d/init.d/asterisk startStarting Asterisk PBX: FATAL: Module ixj not found.

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[asterisk-users] create_addr: No such host:


I have created a context in extensions.conf and when I dial, it is suppose to ask me to enter pin number but instead this the error I get.

Sep 18 18:11:54 WARNING[6514]: chan_sip.c:1968 create_addr: No such host: 4035Sep 18 18:11:54 NOTICE[6514]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)

Below is the channel information in extensions.conf
[applicationxyz];CallingCard applicationexten = 4035,1,Answerexten = 4035,2,Wait,2exten = 4035,3,DeadAGI,applicationxyz.phpexten = 4035,4,Wait,2exten = 4035,5,Hangup

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Re: [asterisk-users] Starting Asterisk PBX: FATAL: Module ixj not found.


The only load I have is,

load = chan_modem.soload = res_musiconhold.so
[global]chan_modem.so=yes

-- Original message -- From: "Justin Tunney" [EMAIL PROTECTED]  Check /etc/asterisk/modules.conf and see if there is a line trying to load it.   On 9/18/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]>wrote:   [EMAIL PROTECTED] asterisk]# /etc/rc.d/init.d/asterisk start   Starting Asterisk PBX: FATAL: Module ixj not found.  ___  --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 


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Re: [asterisk-users] Termination Rates


Thanks I'll give them a trial.

-- Original message -- From: "Insider KT" [EMAIL PROTECTED]  I've used this company now for over a year.  It is part of Ipcb.net, so you got live support 24 hours a day every day.  The quality is very good and the reliability is near perfect. You can have  1000 simultaneous calls.   On the down side - The Signup is not so easy. I had to fax 7 papers to  verify my account. And had to wait a couple of days for them to check it.  You also have to be a company to register and send them your bank  information.   FredrikI saw this termination company, www.BuyMin.com   the   rates looks good. Has anyone any experience with this company? I use   Gafachi, very reliable 
 but ex
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[asterisk-users] Asterisk Server Down

2006-09-17 Thread broadbandvoice


I rebooted the server on which the Asterisk is hosted on. The * did not come back up and I get this message when I attempt to use CLI

[EMAIL PROTECTED] ~]# asterisk -rUnable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)

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Re: [asterisk-users] Asterisk Server Down

2006-09-17 Thread broadbandvoice


Thanks everyone it is working now.

-- Original message -- From: Tzafrir Cohen [EMAIL PROTECTED]  On Sun, Sep 17, 2006 at 10:40:16AM -0400, Steve Totaro wrote:you're right, one should proof, under which user asterisk runs...   Besides security reasons, running asterisk as root, doesn't it allow a   higher prioritization of asterisk processes?   This is why we let asterisk setuid itself to user asterisk, and don't  let the wrappr script handle that. Asterisk sets scheduling priority  before running setuid/setgid .I can see a problem with security issues but is it a bad thing to allow   higher priority of the asterisk process? Not sure that it does anyways,   but I don't see how that is a bad thing?   It can help the qu
 ality 
of Audio. On the downside, it means that a 100%  CPU loop in asterisk is a pain to recover from. Security implications:  if someone can inject you one line to the dialpan, they can (under the  right circumstances) get your system stuck very badly . Unless you have  a manager connection availble.   --  Tzafrir Cohen sip:[EMAIL PROTECTED]  icq#16849755 iax:[EMAIL PROTECTED]  +972-50-7952406 jabber:[EMAIL PROTECTED]  [EMAIL PROTECTED] http://www.xorcom.com  ___  --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 

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[asterisk-users] Termination Rates

2006-09-17 Thread broadbandvoice


I saw this termination company, www.BuyMin.comthe rates looks good. Has anyone any experience with this company? I use Gafachi, very reliable but expensive.

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Re: [asterisk-users] University switches to Asterisk

2006-09-13 Thread broadbandvoice


Thats good news for us.

-- Original message -- From: Doug Lytle [EMAIL PROTECTED]  Interesting article I found linked from Groklaw:   "Sam Houston State University replaces Cisco CallManagers, Nortel PBXs  with Linux-based VoIP and messaging servers"   http://www.networkworld.com/news/2006/091206-von-sam-houston.html?page=1   Doug   --   Ben Franklin quote:   "Those who would give up Essential Liberty to purchase a little Temporary  Safety, deserve neither Liberty nor Safety."___  --Bandwidth and Colocation provided by Easynews.com --   asterisk-users mailing list  To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo
 /aster
isk-users 

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Re: [asterisk-users] Verify Database Installation

2006-09-12 Thread broadbandvoice


This is a questions about database verification and not a2billing. Asterisk also uses database for such things as cdr and sometimes you call dial plans from database. Someone might have seen a similar situation while installing postgres for Asterisk. It is Asterisk related. 

-- Original message -- From: "Areski K" [EMAIL PROTECTED]  Please try to redirect those questions to the appropriate place,  I mean the A2Billing forum : http://forum.asterisk2billing.org  It's off-topics for the Asterisk-user mailing-list.   Kind regards,  /AreskiOn 9/11/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]>wrote:   Everything was going well, I got the tarball, unpacked the tarballs, created   the postgre user and password, database is created and checked ownership and   even got a list of database users. I even imported the data schema into the   new database. My problem now is verification of database installation. I get   an error below when i try it:  
 t;   a2billing= SELECT * FROM cc_ui_authen;   ERROR: relation "cc_ui_authen" does not exist   a2billing=   -- Original message --   From: [EMAIL PROTECTED] You're right. How did I miss that? -- Original message --   From: [EMAIL PROTECTED] Yes, I put the filename   "Asterisk2Billing_release_Chameleon_v1_2_3.tar.gz" and got   that error. -- Original message --   From: "Jamin W. Collins" <[EMAIL PROTECTED]> [EMAIL PROTECTED] wrote: I was successful in getting the tarball for a2billing [EMAIL PROTECTED] a2billing]# ls -all  &
 gt; 
t;  total 4872 drwxr-xr-x 2 root root 4096 Sep 11 06:22 . drwxr-xr-x 20 root root 4096 Sep 10 21:28 .. -rw-r--r-- 1 root root 165 Sep 11 06:16   download.php?get=Asterisk2Billing_release_Chameleon_beta.tar.gz -rw-r--r-- 1 root root 4960345 Sep 11 06:31   download.php?get=Asterisk2Billing_release_Chameleon_v1_2_3.tar.gz  ^The above is your file name, note the additional "download.php?get=" onthe file name.   --   a mp;a m p;g t ; Jam in W. Collins___--Bandwidth and Colocation provided by Easynews.com --  
  aster
isk-users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users -- Forwarded message --   From: [EMAIL PROTECTED]   To: Asterisk Users Mailing List - Non-Commercial Discussion     Date: Mon, 11 Sep 2006 14:23:43 +   Subject: Re: [asterisk-users] Problems Unpacking tarball For Asterisk   Application   ___   --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 -- Forwarded message -- 
 
t;  From: [EMAIL PROTECTED]   To: Asterisk Users Mailing List - Non-Commercial Discussion     Date: Mon, 11 Sep 2006 14:17:25 +   Subject: Re: [asterisk-users] Problems Unpacking tarball For Asterisk   Application   ___   --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list   To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users   ___   --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list   To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/ast
 erisk-
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Re: [asterisk-users] sip origination and termination

2006-09-12 Thread broadbandvoice


You're right Voxee support sucks. But I think they do well and provide good rates. I'm using Gafachi, a little expensive and have Voxee. I'm using LCR so the termination will try Voxee first and when not available will use Gafachi. You can set up something like that with a least cost routing.

-- Original message -- From: Christopher Corn [EMAIL PROTECTED] 
im finding companies like voxee that offer very low rates and then companies like voipstreet that offer at a higher rate doube. whats the catch? is voxee, what you would call a wholesaler and voipstreet, commercial?

im worried about going with companies like voxeee, because i question their support. 

what are your guys thoughts on this? Thanks.

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Re: [asterisk-users] su - postgres -bash-3.00$


Thanks for all the help from Collin, I figured out the postgres database creation. I now have a different problem trying download files. I get this error. Maybe wget does not work in Fedora.

[EMAIL PROTECTED] a2billing]# wget http://www.areski.net/Open_A2Billing_version_Racoon.tar.gz--05:52:17-- http://www.areski.net/Open_A2Billing_version_Racoon.tar.gz = `Open_A2Billing_version_Racoon.tar.gz'Resolving www.areski.net... 213.186.33.19Connecting to www.areski.net|213.186.33.19|:80... connected.HTTP request sent, awaiting response... 404 Not Found05:52:17 ERROR 404: Not Found.
[EMAIL PROTECTED] a2billing]# 

-- Original message -- From: Collin Baillie [EMAIL PROTECTED]  [EMAIL PROTECTED] wrote:   Yes it is running postgres 4049 0.0 0.3 20360 3072 ? S May24 0:00   /usr/bin/postmaster -p 5432 -D /var/lib/pgsql/data   Alright! So if you can do this:   [EMAIL PROTECTED] /]# su - postgres  [bash-3.00 ~]$ whoami   It should return something like:   postgres   or if you can do:   [bash-3.00 ~]$ id -a   it should return something like:   uid=5000(postgres) gid=5000(postgres) groups=5000(postgres)   Collin 

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Re: [asterisk-users] su - postgres -bash-3.00$


I figured it out, I had old install manual.

-- Original message -- From: [EMAIL PROTECTED] 
Thanks for all the help from Collin, I figured out the postgres database creation. I now have a different problem trying download files. I get this error. Maybe wget does not work in Fedora.

[EMAIL PROTECTED] a2billing]# wget http://www.areski.net/Open_A2Billing_version_Racoon.tar.gz--05:52:17-- http://www.areski.net/Open_A2Billing_version_Racoon.tar.gz = `Open_A2Billing_version_Racoon.tar.gz'Resolving www.areski.net... 213.186.33.19Connecting to www.areski.net|213.186.33.19|:80... connected.HTTP request sent, awaiting response... 404 Not Found05:52:17 ERROR 404: Not Found.
[EMAIL PROTECTED] a2billing]# 

-- Original message -- From: Collin Baillie [EMAIL PROTECTED]  [EMAIL PROTECTED] wrote:   Yes it is running postgres 4049 0.0 0.3 20360 3072 ? S May24 0:00   /usr/bin/postmaster -p 5432 -D /var/lib/pgsql/data   Alright! So if you can do this:   [EMAIL PROTECTED] /]# su - postgres  [bash-3.00 ~]$ whoami   It should return something like:   postgres   or if you can do:   [bash-3.00 ~]$ id -a   it should return something like:   uid=5000(postgres) gid=5000(postgres) groups=5000(postgres)   Collin 

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[asterisk-users] Problems Unpacking tarball For Asterisk Application


I was successful in getting the tarball for a2billing

[EMAIL PROTECTED] a2billing]# ls -alltotal 4872drwxr-xr-x 2 root root 4096 Sep 11 06:22 .drwxr-xr-x 20 root root 4096 Sep 10 21:28 ..-rw-r--r-- 1 root root 165 Sep 11 06:16 download.php?get=Asterisk2Billing_release_Chameleon_beta.tar.gz-rw-r--r-- 1 root root 4960345 Sep 11 06:31 download.php?get=Asterisk2Billing_release_Chameleon_v1_2_3.tar.gz

But I'm having trouble unpacking the tarballs: Below is the command and error I get:
[EMAIL PROTECTED] a2billing]# tar -xzvf Asterisk2Billing_release_Chameleon_v1_2_3.tar.gztar: Asterisk2Billing_release_Chameleon_v1_2_3.tar.gz: Cannot open: No such file or directorytar: Error is not recoverable: exiting nowtar: Child returned status 2tar: Error exit delayed from previous errors
I triedchanging the directory to -rw-r--r-- but got a different error

[EMAIL PROTECTED] a2billing]# tar -rw-r--r-- Asterisk2Billing_release_Chameleon_v1_2_3.taz.gztar: --: (PROGRAM ERROR) Option should have been recognized!?Try `tar --help' or `tar --usage' for more information.[EMAIL PROTECTED] a2billing]# 

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Re: [asterisk-users] Problems Unpacking tarball For Asterisk Application


Yes, I put the filename "Asterisk2Billing_release_Chameleon_v1_2_3.tar.gz" and got that error.

-- Original message -- From: "Jamin W. Collins" [EMAIL PROTECTED]  [EMAIL PROTECTED] wrote:   I was successful in getting the tarball for a2billing [EMAIL PROTECTED] a2billing]# ls -all   total 4872   drwxr-xr-x 2 root root 4096 Sep 11 06:22 .   drwxr-xr-x 20 root root 4096 Sep 10 21:28 ..   -rw-r--r-- 1 root root 165 Sep 11 06:16   download.php?get=Asterisk2Billing_release_Chameleon_beta.tar.gz   -rw-r--r-- 1 root root 4960345 Sep 11 06:31   download.php?get=Asterisk2Billing_release_Chameleon_v1_2_3.tar.gz  ^  The above is your file name, note the additional "download.php?get=" on  the file name.   -- 
 t; Jam
in W. Collins  ___  --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 

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Re: [asterisk-users] Problems Unpacking tarball For Asterisk Application


You're right. How did I miss that?

-- Original message -- From: [EMAIL PROTECTED] 
Yes, I put the filename "Asterisk2Billing_release_Chameleon_v1_2_3.tar.gz" and got that error.

-- Original message -- From: "Jamin W. Collins" [EMAIL PROTECTED]  [EMAIL PROTECTED] wrote:   I was successful in getting the tarball for a2billing [EMAIL PROTECTED] a2billing]# ls -all   total 4872   drwxr-xr-x 2 root root 4096 Sep 11 06:22 .   drwxr-xr-x 20 root root 4096 Sep 10 21:28 ..   -rw-r--r-- 1 root root 165 Sep 11 06:16   download.php?get=Asterisk2Billing_release_Chameleon_beta.tar.gz   -rw-r--r-- 1 root root 4960345 Sep 11 06:31   download.php?get=Asterisk2Billing_release_Chameleon_v1_2_3.tar.gz  ^  The above is your file name, note the additional "download.php?get=" on  the file name.   -- 
 mp;g t
; Jam in W. Collins  ___  --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 

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[asterisk-users] Verify Database Installation


Everything was going well, I got the tarball, unpacked the tarballs, created the postgre user and password, database is created and checked ownership and even got a list of database users. I even imported the data schema into the new database. My problem now is verification of database installation. I get an error below when i try it:

a2billing= SELECT * FROM cc_ui_authen;ERROR: relation "cc_ui_authen" does not exista2billing= 

-- Original message -- From: [EMAIL PROTECTED] 
You're right. How did I miss that?

-- Original message -- From: [EMAIL PROTECTED] 
Yes, I put the filename "Asterisk2Billing_release_Chameleon_v1_2_3.tar.gz" and got that error.

-- Original message -- From: "Jamin W. Collins" [EMAIL PROTECTED]  [EMAIL PROTECTED] wrote:   I was successful in getting the tarball for a2billing [EMAIL PROTECTED] a2billing]# ls -all   total 4872   drwxr-xr-x 2 root root 4096 Sep 11 06:22 .   drwxr-xr-x 20 root root 4096 Sep 10 21:28 ..   -rw-r--r-- 1 root root 165 Sep 11 06:16   download.php?get=Asterisk2Billing_release_Chameleon_beta.tar.gz   -rw-r--r-- 1 root root 4960345 Sep 11 06:31   download.php?get=Asterisk2Billing_release_Chameleon_v1_2_3.tar.gz  ^  The above is your file name, note the additional "download.php?get=" on  the file name.   -- 
 mp;a m
p;g t ; Jam in W. Collins  ___  --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 

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[asterisk-users] su - postgres -bash-3.00$

2006-09-10 Thread broadbandvoice

I'm trying to install Asterisk billing server and when I put in su - postgres I 
get this response instead of the password response.
-bash-3.00$
-bash-3.00$
-bash-3.00$ 

Anyone seen this before? I'm using Fedora core 4 and have the same on a local 
machine that works fine.
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Re: [asterisk-users] can someone recommend a voip provider that...

2006-09-10 Thread broadbandvoice

Better to get one provider that does origination and termination and has no minimum requirements. Most companies will require a deposit or minimum usage requirements. Make sure for origination you know the diffference between metered and unmetered DIDs. codec 729 will be pushing it a little bit. It should not really matter since the termination and origination will take place on their servers and they'll deal with the bandwidth issues. All you will do is modify your sip.conf to hand over the traffic or better yet, some of them will register your phones on their server and in that case you need to do nothing but provision your phones based on their ip and authentication. I hope that helps.

-- Original message -- From: Christopher Corn [EMAIL PROTECTED] ok maybe thats asking for too much. how about a voip provider that provides 729 codec support ? :)Christopher Corn [EMAIL PROTECTED] wrote:

offers unlimited calls, in and out in the US
asterisk support
no setup fee
and support 729 codec?
and of course is reliable and clear

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Re: [asterisk-users] using residential voip for business?

2006-09-10 Thread broadbandvoice


Its a trickish business, when they say unlimited and you make more than 2500 minutes they cut you off.

-- Original message -- From: Christopher Corn [EMAIL PROTECTED] 
I spoke to a voip provider today who mentioned that though they offer an unlimited plan, if we use it for a business and it is over-utilized, it will be canceled.

is this true for all residential voip plans? i have a small office of about 4 or 5 phones. i tend to chose residential plans because they have the unlimited offer for outgoing/incoming.

thx

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Re: [asterisk-users] Call Forwarding in SIP.conf


Thanks all. It works fine now.

-- Original message -- From: "Tim St. Pierre" [EMAIL PROTECTED]  ___  --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 

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Check your Dial() string to make sure that you haven't mistyped and put 
gafachi-o instead of gafachi-out.  Specifiying the full host name will also 
work.

As a hint, you can refresh these changes with out restarting your server (and 
therefore without disrupting any calls in progress)

extensions reload will refresh the extensions file
reload will reload all your configs
sip reload will reload only sip configs (and re-register everything)

Very handy when working on an active machine.


On September 8, 2006 14:19, [EMAIL PROTECTED] wrote:
 Tim, this is the way I have Gafachi set up in sip.conf and works well with
 channels that have an ATA attached to it but not the virtual one. I have
 changed the host in extensions.conf to the .sip.gafachi.com.
 But I have calls on the server and cannot restart it yet. I'll keep you
 posted and thanks for the feedback.

 [gafachi-out]
 type=peer
 secret=xx
 username=x
 fromuser=x
 fromdomain=xxx
 host=.sip.gafachi.com
 ;usereqphone=yes; This provider requires ;user=phone on
 URI ;nat=yes
 rtptimeout=60
 dtmfmode=rfc2833


 -- Original message --
 From: Tim St. Pierre [EMAIL PROTECTED]

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Tim St. Pierre

IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]


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[asterisk-users] Call Processing Slow 11 seconds


I'm having some slowness issue with Asterisk. When a number is dialed it takes 11 seconds before it rings out. I been considering using openser for the call processing and leaving asterisk for voicemail and conference bridge. I get a dialtone rightaway when the receiver is picked up but after dialing the number but within asterisk extensions and pstn numbers takes 11 seconds before ringing out. Anyone else experiencing this. I use Asterisk 1.2.3

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[asterisk-users] DID Provider in Thailand


Does anyone know of a DID provider in Thailand?

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Re: [asterisk-users] Call Forwarding in SIP.conf

I have a follow up question. How do I pass on the caller ID of the call I'm forwarding to the other party? I can pass on the channels caller ID but prefer to pass on the forwarding party's number instead.

-- Original message -- From: [EMAIL PROTECTED]
Thanks all. It works fine now.

-- Original message -- From: "Tim St. Pierre" [EMAIL PROTECTED] ___ --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

---BeginMessage---
---BeginMessage---
Check your Dial() string to make sure that you haven't mistyped and put
gafachi-o instead of gafachi-out. Specifiying the full host name will also
work.

As a hint, you can refresh these changes with out restarting your server (and
therefore without disrupting any calls in progress)

extensions reload will refresh the extensions file
reload will reload all your configs
sip reload will reload only sip configs (and re-register everything)

Very handy when working on an active machine.

On September 8, 2006 14:19, [EMAIL PROTECTED] wrote:
Tim, this is the way I have Gafachi set up in sip.conf and works well with
channels that have an ATA attached to it but not the virtual one. I have
changed the host in extensions.conf to the .sip.gafachi.com.
But I have calls on the server and cannot restart it yet. I'll keep you
posted and thanks for the feedback.

[gafachi-out]
type=peer
secret=xx
username=x
fromuser=x
fromdomain=xxx
host=.sip.gafachi.com
;usereqphone=yes; This provider requires ;user=phone on
URI ;nat=yes
rtptimeout=60
dtmfmode=rfc2833

-- Original message --
From: Tim St. Pierre [EMAIL PROTECTED]

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sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]

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Re: [asterisk-users] Call Forwarding in SIP.conf


I tried both of them but it still goes asID unavailable. First I commented it out, that did not work and left it blank and that did not work either. Below is the sample in sip.conf

[4305]type=frienduser=4305secret=xxx;context=from-sipcallerid= ; left it blank but did not get passed on!host=dynamicnat=yesqualify=yescanreinvite=nodtmfmode=rfc2833

-- Original message -- From: "Tim St. Pierre" [EMAIL PROTECTED]  ___  --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 

---BeginMessage---
If you don't set the callerID in the channel, it will get passed on as-is.  
Don't change it, and it will stay the same.

-TIm

On September 9, 2006 12:27, [EMAIL PROTECTED] wrote:
 I have a follow up question. How do I pass on the caller ID of the call I'm
 forwarding to the other party? I can pass on the channels caller ID but
 prefer to pass on the forwarding party's number instead.

 -- Original message --
 From: [EMAIL PROTECTED]

 Thanks all. It works fine now.

 -- Original message --
 From: Tim St. Pierre [EMAIL PROTECTED]

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-- 
Tim St. Pierre

IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]


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RE: [asterisk-users] Call Processing Slow 11 seconds

Yes that works. I'm using Linksys adapter, is there a code I can put in the dial plan to prevent users from putting # after the number? I have a lot of people on the server and cannot ask them all to be pushing # after every call. Thanks for the tip and any help will be appreciated.

-- Original message -- From: "G.Jacobsen" [EMAIL PROTECTED]

In case you use an adapter or voip phone: Did you try to press hash # after the number ? - thenthe adapter/voip phonedials immediately and doesnt wait for the next digit timeout.

Cheers

Gerry

-Original MessageFrom: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of [EMAIL PROTECTED]Sent: Samstag, 9. September 2006 15:15To: asterisk-users@lists.digium.comSubject: [asterisk-users] Call Processing Slow 11 seconds
I'm having some slowness issue with Asterisk. When a number is dialed it takes 11 seconds before it rings out. I been considering using openser for the call processing and leaving asterisk for voicemail and conference bridge. I get a dialtone rightaway when the receiver is picked up but after dialing the number but within asterisk extensions and pstn numbers takes 11 seconds before ringing out. Anyone else experiencing this. I use Asterisk 1.2.3

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Re: [asterisk-users] Call Processing Slow 11 seconds

Thanks, I tried that and did not work for me. My users are calling US number and without the # at the end of the last digit dials it takes 11 seconds before it starts ringing.

-- Original message -- From: Alberto Sagredo [EMAIL PROTECTED] Yes you could script a dialplan putting ... and S0 (zero) at the end. An example : (xxS0) It will dial 6 digits directly when you enter the 6th. You could learn how to adapt your Linksys dialplan looking this wiki. http://voip.wikispaces.com/ [EMAIL PROTECTED] escribió: Yes that works. I'm using Linksys adapter, is there a code I can put in the dial plan to prevent users from putting # after the number? I have a lot of people on the server and cannot ask them all to be pushing # after every call. Thanks for the tip and any help will be appreciated. --
O
riginal message -- From: "G.Jacobsen" <[EMAIL PROTECTED]> In case you use an adapter or voip phone: Did you try to press hash # after the number ? - then the adapter/voip phone dials immediately and doesnt wait for the next digit timeout. Cheers Gerry -Original Message *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] *Sent:* Samstag, 9. September 2006 15:15 *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Call Processing Slow 11 seconds I'm having some slowness issue with Asterisk. When a number is dialed it takes 11 seconds before it rings out. I been conside
ring u
sing openser for the call processing and leaving asterisk for voicemail and conference bridge. I get a dialtone rightaway when the receiver is picked up but after dialing the number but within asterisk extensions and pstn numbers takes 11 seconds before ringing out. Anyone else experiencing this. I use Asterisk 1.2.3 Asunto: RE: [asterisk-users] Call Processing Slow 11 seconds De: "G.Jacobsen" <[EMAIL PROTECTED]> Fecha: Sat, 9 Sep 2006 17:20:05 + Para: "Asterisk Users Mailing List - Non-Commercial Discussion" Para: "Asterisk Users Mailing List - Non-Commercial Discussion"
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st To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] Whcih phones are better for mass deployment


Try the Linksys ATA. I gave up on Granstream and have 4 sitting in around.

-- Original message -- From: "Zeeshan Zakaria" [EMAIL PROTECTED] 
I am having hard time with grandstream phones for a30 phone setup. When a change in configuration is required, I have to change their configurations manually for almost all of them. Their configuration utility is not very straight forward to use. 

For my next installation, I would prefer some other phones with better configuration and remote accress utility. My question to those of you with more experience, what IP phones are better for mass deployment and easy management of updates and configurations? Or what other solution is better for mass deployment of phones? -- Zeeshan A Zakaria 

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Re: [asterisk-users] Whcih phones are better for mass deployment


for mass deployment the Linksys will allow you to update your routers with a tftp server.. You can have the routers always download their software from the tftp server, that way you have the latest on the server for upgrade software. The reason that I don't like granstream is their bad customer support, they live you on your own basically.

-- Original message -- From: "Zeeshan Zakaria" [EMAIL PROTECTED] Can you explain a little bit what make them better for mass deployment. Do they have Windows based software to communicate with all the installed phones and upgrade them and also to remotely monitor them. Is there a separate cost for these software tools or are they free? 

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Re: [asterisk-users] Call Forwarding in SIP.conf

2006-09-08 Thread broadbandvoice


It sounds like a good idea, I tried it and get this error


Sep 8 09:52:17 WARNING[27193]: chan_sip.c:1968 create_addr: No such host: gafachi-o
Sep 8 09:52:17 NOTICE[27193]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)

In Extensions.conf I have
exten = 4305,1,Dial(SIP/[EMAIL PROTECTED]) ; permit transfer

In Sip.conf I have
[4305]
type=friend
user=4305
secret=
;context=from-sip
callerid=
host=dynamic
nat=yes
canreinvite=no
dtmfmode=rfc2833
;incominglimit=1
;[EMAIL PROTECTED]
;disallow=all
;allow=ulaw
;allow=alaw
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!



-- Original message -- From: "William Piper" [EMAIL PROTECTED] 
whatever the did is needs to be put in the extensions.conf  told to dial your cellphone. 
Example:

exten = _011123445566,1,Dial,SIP/[EMAIL PROTECTED] 

assuming that your using a SIP carrier, replace 1234567890 with your cellphone  1.2.3.4 with the carrier's IP or carriers context name in sip.conf.
bp
On 9/8/06, [EMAIL PROTECTED] [EMAIL PROTECTED]  wrote: 



I'm using it for virtual numbers. I have international virtual number from a DID provider and want to forward it to my cell phone.

In Sip.conf I have the channel


[4305]
type=friend
user=4305
secret=
;context=from-sip
callerid=
host=dynamic
nat=yes
canreinvite=no
dtmfmode=rfc2833
;incominglimit=1
;[EMAIL PROTECTED]
;disallow=all
;allow=ulaw
;allow=alaw
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!

and in extensions.conf I have

exten = 4305,1,Dial(SIP/4305,120,rt) ; permit transfer

This had worked in the past when I forwarded it through the Linksys ATA but now have run out of ATA's.


-- Original message -- From: "Tim St. Pierre"  [EMAIL PROTECTED]  ___  --Bandwidth and Colocation provided by Easynews.com --   asterisk-users mailing list  To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] Call Forwarding in SIP.conf

2006-09-08 Thread broadbandvoice


Tim, this is the way I have Gafachi set up in sip.conf and works well with channels that have anATA attached to it but not the virtual one. I have changed the host in extensions.conf to the .sip.gafachi.com. But I have calls on the server and cannot restart it yet. I'll keep you posted and thanks for the feedback.

[gafachi-out]type=peersecret=xxusername=xfromuser=xfromdomain=xxxhost=.sip.gafachi.com;usereqphone=yes ; This provider requires ";user=phone" on URI;nat=yesrtptimeout=60dtmfmode=rfc2833

-- Original message -- From: "Tim St. Pierre" [EMAIL PROTECTED]  ___  --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 

---BeginMessage---
Do you have gafachi-o in your sip.conf?

Since it's not a valid host name, you need to have an entry in sip.conf to 
tell asterisk how to make a call to gafachi-o.

That's why it is telling you No such host.



On September 8, 2006 12:57, [EMAIL PROTECTED] wrote:
 It sounds like a good idea, I tried it and get this error

 Sep  8 09:52:17 WARNING[27193]: chan_sip.c:1968 create_addr: No such host:
 gafachi-o Sep  8 09:52:17 NOTICE[27193]: app_dial.c:1011 dial_exec_full:
 Unable to create channel of type 'SIP' (cause 3 - No route to destination)

 In Extensions.conf I have
 exten = 4305,1,Dial(SIP/[EMAIL PROTECTED])  ; permit transfer

 In Sip.conf I have
 [4305]
 type=friend
 user=4305
 secret=
 ;context=from-sip
 callerid=
 host=dynamic
 nat=yes
 canreinvite=no
 dtmfmode=rfc2833
 ;incominglimit=1
 ;[EMAIL PROTECTED]
 ;disallow=all
 ;allow=ulaw
 ;allow=alaw
 ;allow=g723.1   ; Asterisk only supports g723.1 pass-thru!



 -- Original message --
 From: William Piper [EMAIL PROTECTED]

 whatever the did is needs to be put in the extensions.conf  told to dial
 your cellphone. Example:

 exten = _011123445566,1,Dial,SIP/[EMAIL PROTECTED]

 assuming that your using a SIP carrier, replace 1234567890 with your
 cellphone  1.2.3.4 with the carrier's IP or carriers context name in
 sip.conf.

 bp

 On 9/8/06, [EMAIL PROTECTED] [EMAIL PROTECTED]  wrote:
 I'm using it for virtual numbers. I have international virtual number from
 a DID provider and want to forward it to my cell phone.

 In Sip.conf I have the channel

 [4305]
 type=friend
 user=4305
 secret=
 ;context=from-sip
 callerid=
 host=dynamic
 nat=yes
 canreinvite=no
 dtmfmode=rfc2833
 ;incominglimit=1
 ;[EMAIL PROTECTED]
 ;disallow=all
 ;allow=ulaw
 ;allow=alaw
 ;allow=g723.1   ; Asterisk only supports g723.1 pass-thru!

 and in extensions.conf I have

 exten = 4305,1,Dial(SIP/4305,120,rt)  ; permit transfer

 This had worked in the past when I forwarded it through the Linksys ATA but
 now have run out of ATA's.


 -- Original message --
 From: Tim St. Pierre  [EMAIL PROTECTED]

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-- 
Tim St. Pierre

IP telephony specialist
sip://[EMAIL PROTECTED]
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
[EMAIL PROTECTED]


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[asterisk-users] Call Forwarding in SIP.conf

2006-09-07 Thread broadbandvoice


I looked through the forums but could not find exactly what I needed. I need help setting up call forwarding in sip.conf, where the call forwards to PSTN number without a sip phone but just the channels in sip.conf without any hardware or softphone. Any help will be greatly appreciated.

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Re: [asterisk-users] Call Forwarding in SIP.conf

2006-09-07 Thread broadbandvoice


I'm using it for virtual numbers. I have international virtual number from a DID provider and want to forward it to my cell phone.

In Sip.conf I have the channel


[4305]
type=friend
user=4305
secret=
;context=from-sip
callerid=
host=dynamic
nat=yes
canreinvite=no
dtmfmode=rfc2833
;incominglimit=1
;[EMAIL PROTECTED]
;disallow=all
;allow=ulaw
;allow=alaw
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!

and in extensions.conf I have

exten = 4305,1,Dial(SIP/4305,120,rt) ; permit transfer

This had worked in the past when I forwarded it through the Linksys ATA but now have run out of ATA's.


-- Original message -- From: "Tim St. Pierre" [EMAIL PROTECTED]  ___  --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 

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[asterisk-users] ERROR 1045 (28000): Access denied for user

2006-07-24 Thread broadbandvoice


MySql password for root:
Domain (realm) for the default user 'admin': localhost.localdomain

creating database openser ...
ERROR 1045 (28000): Access denied for user 'root'@'localhost' (using password: 
Y ES)
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RE: [asterisk-users] Asterisk Dial Plan to Play Message

2006-07-23 Thread broadbandvoice


It did not work, how can I put in some user intervention so that any numbers they dial will send them to a message? Restrict their outbound calls and a get a message to contact administrator instead of a busy signal.

-- Original message -- From: "brandon kruz" [EMAIL PROTECTED]  thank you russel  forgot to mention this.From: Russell Bryant <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial  Discussion  To: Asterisk Users Mailing List - Non-Commercial  Discussion  Subject: RE: [asterisk-users] Asterisk Dial Plan to Play Message  Date: Sat, 22 Jul 2006 21:29:23 -0400  MIME-Version: 1.0  Received: from lists.digium.com ([69.16.138.164]) by  bay0-mc9-f18.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Sat,  22 Jul 2006 18:32:43 -0700  Received: from digium-69-16-138-164.phx1.puregig.net (
 localh
ost  [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 563422FC409;Sat, 22  Jul 2006 18:29:35 -0700 (MST)  Received: from abita.digium.internal (gateway.digium.com [216.207.245.1])by  lists.digium.com (Postfix) with ESMTP id 9CD5C2FC25Cfor  ;Sat, 22 Jul 2006 18:29:24 -0700 (MST)  Received: from heineken.digium.com (heineken.digium.internal [10.16.1.2])by  abita.digium.internal (Postfix) with ESMTP id C536AA94939for  ;Sat, 22 Jul 2006 20:29:25 -0500 (CDT)  Received: from [172.17.99.18] ([172.17.99.18])by heineken.digium.com  (8.13.6/69.69.69) with ESMTP id k6N1UIr9030086for  ; Sat, 22 Jul 2006 20:30:19 -0500  X-Message-Info: LsUYwwHHNt14xbUYi+9bCaWgpoxRQZbXIFwSWMVl+QA=  X-Original-To: asterisk-users@lists.digium.com 
  &
gt;Delivered-To: asterisk-users@lists.digium.com  References: <[EMAIL PROTECTED]> Organization: Digium, Inc.  X-Mailer: Evolution 2.6.1 X-BeenThere: asterisk-users@lists.digium.com  X-Mailman-Version: 2.1.5  Precedence: list  List-Id: Asterisk Users Mailing List - Non-Commercial  Discussion  List-Unsubscribe:  , [EMAIL PROTECTED]  List-Archive:  List-Post:  List-Help:  List-Subscribe:  , [EMAIL PROTECTED]
 igium.
com?subject=subscribe  Errors-To: [EMAIL PROTECTED]  Return-Path: [EMAIL PROTECTED]  X-OriginalArrivalTime: 23 Jul 2006 01:32:44.0557 (UTC)  FILETIME=[E58083D0:01C6ADF7]On Sat, 2006-07-22 at 19:38 -0500, brandon kruz wrote:[internal]exten = s,1,Answer()exten = s,n,Playback(custom)exten = s,n,Hangup()This, by itself, does not solve the problem where you want the message  to be played back when the phone is picked up without any user  intervention. If you're using zap phones, you can simply set this  option:immediate=yesThen, as soon as the phone goes off hook, the call will begin at the 's'  extension in the configured context instead 
 of pro
viding dialtone.--  Russell Bryant  Software Developer  Digium, Inc.___  --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   _  Dont just search. Find. Check out the new MSN Search!  http://search.msn.click-url.com/go/onm00200636ave/direct/01/   ___  --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 

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Re: [asterisk-users] Asterisk Dial Plan to Play Message

2006-07-23 Thread broadbandvoice


Thanks for the response, its looks logical, for some reason the authentication is not working for me, I'm using a Linksys Phone adapter and here is a sample dial plan in extensions.conf and also SIP channels.

exten = 8407,1,Dial(SIP/8407,80,rt) ; permit transferexten = 8407,n,Authenticate(9461) exten = 8407,n,Playback(pbx-invalid)exten = 8407,n,Hangup()

and in sip.conf

[9507]type=frienduser=8407secret=xx;context=from-sipcallerid=8407host=dynamicnat=yesqualify=yescanreinvite=nodtmfmode=rfc2833;incominglimit=1;[EMAIL PROTECTED];disallow=all;allow=ulaw;allow=alaw;allow=g729;allow=g723.1
I also tried changing type to peer instead of freind and does not work either. I am running Asterisk 1.2.3.

Any help will be appreciated and thanks for all your inputs.


-- Original message -- From: Steve Totaro [EMAIL PROTECTED]  If you are using phones attached to a ZAP FXS port the immediate=yes  will work. Otherwise, some SIP phones (Grandstream for instance) allows  you to enter an autodial number. It depends on what is providing the  dialtone to the handset. If your device does not support autodial, then  the next best thing is to do what has already been suggested.   OR   [somecontext]  exten=s,1,answer  exten=s,2,Authenticate(insertdigitshere)  exten=s,3,(continue with a real dialplan)   Change the corresponding authenticate gsm file to say what you want  about contacting the boss.   This gives the impression that phone is restricted for outbound calling  b
 ut if 
you enter the authenticate string, you can dialout for  emergencies or convenience.   Thanks,  Steve   [EMAIL PROTECTED] wrote:   It did not work, how can I put in some user intervention so that any   numbers they dial will send them to a message? Restrict their outbound   calls and a get a message to contact administrator instead of a busy   signal. -- Original message --   From: "brandon kruz" <[EMAIL PROTECTED]> thank you russelforgot to mention this.   On Sat, 2006-07-22 at 19:38 -0500, brandon kruz wrote:  [internal]  exten = s,1,Answer()  exten = s,n,Playback(custom) 
  exten
 = s,n,Hangup()This, by itself, does not solve the problem where you want the   messageto be played back when the phone is picked up without any userintervention. If you're using zap phones, you can simply set thisoption:immediate=yesThen, as soon as the phone goes off hook, the call will begin   at the 's'extension in the configured context instead of pro viding   dialtone.--Russell BryantSoftware DeveloperDigium, Inc.___--Bandwidth and Colocation provided by Easynews.
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Re: [asterisk-users] Asterisk Dial Plan to Play Message

2006-07-23 Thread broadbandvoice


That was a typo its corrected to [8407] but problem still persist with original questions though.

-- Original message -- From: "Eric "ManxPower" Wieling" [EMAIL PROTECTED]  "[9507]" is the incoming User ID. "user=8407" is the outgoing User ID.  Do you really want them to be different?   Dial() will stop processing of the dialplan until the call ends. Do you  really want this?   "r" option to Dial will force a ringing sound to the caller, even if the  caller should be hearing a "all circuits are busy", or "your call cannot  be completed as dialed" or similar message. Do you really want that?   [EMAIL PROTECTED] wrote:   Thanks for the response, its looks logical, for some reason the authentication  is not working for me, I'm using a Linksys Phone adapter and here is a sample  dial plan in extensions.conf and also SIP channels.
  &
gt;exten = 8407,1,Dial(SIP/8407,80,rt) ; permit transfer   exten = 8407,n,Authenticate(9461)   exten = 8407,n,Playback(pbx-invalid)   exten = 8407,n,Hangup() and in sip.conf [9507]   type=friend   user=8407   secret=xx   ;context=from-sip   callerid=8407   host=dynamic   nat=yes   qualify=yes   canreinvite=no   dtmfmode=rfc2833   --  Now accepting new clients in Birmingham, Atlanta, Huntsville,  Chattanooga, and Montgomery.  ___  --Bandwidth and Colocation provided by Easynews.com --   asterisk-users mailing list  To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk
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[asterisk-users] Asterisk Dial Plan to Play Message

2006-07-22 Thread broadbandvoice


Anyone know how to use dial plan to play messages as soon as a phone is picked up. Like when a user picks up a phone, get a message to contact administrator instead of a dial tone?

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RE: [asterisk-users] Asterisk Dial Plan to Play Message

2006-07-22 Thread broadbandvoice


Thanks, I'll try that in a few hours and share the experience.

-- Original message -- From: "brandon kruz" [EMAIL PROTECTED]  youll have to decide what context this goes in  either  [internal]  or [incoming]  but i hope you can figure this out yourself  here is an idea   [internal]  exten = s,1,Answer()  exten = s,n,Playback(pbx-invalid)  exten = s,n,Hangup()   so now when a user from [internal] picks up the phone the pbx-invalid.gsm is  played from  the asterisk sounds directory  so now you are saying to yourself i want my own voice, to record  check this out   [internal]  exten = s,1,Answer()  exten = s,n,Playback(custom)  exten = s,n,Hangup()   (dont tell anyone this part, or just add it temp
 oraril
y)   exten = 999,1,Answer()  exten = 999,n,Record(custom.gsm)  exten = 999,n,Wait(1)  exten = 999,n,Playback(custom)   you will hear a beep after u dial 999, start recording  then hangup or i think u can type # to stop and move to the next optionFrom: [EMAIL PROTECTED]  Reply-To: Asterisk Users Mailing List - Non-Commercial  Discussion  To: asterisk-users@lists.digium.com  Subject: [asterisk-users] Asterisk Dial Plan to Play Message  Date: Sat, 22 Jul 2006 20:00:37 +  MIME-Version: 1.0  Received: from lists.digium.com ([69.16.138.164]) by  bay0-mc12-f15.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Sat,  22 Jul 2006 13:02:18 -0700  Received: from digium-69-16-138-164.phx1.puregig.net (localhos
 t 
 [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 81F652FC9F5;Sat, 22  Jul 2006 13:00:42 -0700 (MST)  Received: from psmtp.com (exprod8mx22.postini.com [64.18.3.122])by  lists.digium.com (Postfix) with SMTP id DD3DF2FC9BFfor  ;Sat, 22 Jul 2006 13:00:37 -0700 (MST)  Received: from source ([216.148.227.153]) by  exprod8mx22.postini.com([64.18.7.10]) with SMTP; Sat, 22 Jul 2006 13:00:38  PDT  Received: from rmailcenter18.comcast.net ([204.127.197.128])by comcast.net  (rwcrmhc13) with SMTPid 20060722200038m1300da2kae; Sat, 22 Jul 2006  20:00:38 +  Received: from [208.17.34.25] by rmailcenter18.comcast.net;Sat, 22 Jul 2006  20:00:37 +  X-Message-Info: txF49lGdW41NVVYVTXpoQhqJpeo4y5ph0jUyL25Wnpk=  X-Original-To: asterisk-users@lists.digium.com  &
 gt;Del
ivered-To: asterisk-users@lists.digium.com  X-Mailer: ATT Message Center Version 1 (Apr 11 2006)  X-Authenticated-Sender: bnRhbmRvaEBjb21jYXN0Lm5ldA==  X-pstn-levels: (S:99.9/99.9 FC:95.5390 LC:95.5390 R:95.9108  P:95.9108M:96.8350 C:98.4741 )  X-pstn-settings: 3 (1.:1.) s fc lc gt3 gt2 gt1 r p m c  X-pstn-addresses: from <[EMAIL PROTECTED]>[db-null] X-BeenThere:  asterisk-users@lists.digium.com  X-Mailman-Version: 2.1.5  Precedence: list  List-Id: Asterisk Users Mailing List - Non-Commercial  Discussion  List-Unsubscribe:  , [EMAIL PROTECTED]  List-Archive:  List-Post:  List-Help:  List-Subscribe:  , [EMAIL PROTECTED]  Errors-To: [EMAIL PROTECTED]  Return-Path: [EMAIL PROTECTED]  X-OriginalArrivalTime: 22 Jul 2006 20:02:19.0387 (UTC)  FILETIME=[BCC7A4B0:01C6ADC9]Anyone know how to use dial plan to play messages as soon as a phone is  picked up. Like when a user picks up a phone, get a message to contact  administrator instead of a dial tone?___  --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list  To UN
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RE: [asterisk-users] Asterisk Dial Plan to Play Message

2006-07-22 Thread broadbandvoice


I'm using SIP channel in Sip.conf and hand the calls over to a termination vendor. 

-- Original message -- From: Russell Bryant [EMAIL PROTECTED]  On Sat, 2006-07-22 at 19:38 -0500, brandon kruz wrote:   [internal]   exten = s,1,Answer()   exten = s,n,Playback(custom)   exten = s,n,Hangup()   This, by itself, does not solve the problem where you want the message  to be played back when the phone is picked up without any user  intervention. If you're using zap phones, you can simply set this  option:   immediate=yes   Then, as soon as the phone goes off hook, the call will begin at the 's'  extension in the configured context instead of providing dialtone.   --  Russell Bryant  Software Developer  Digium, Inc.   
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[asterisk-users] sip_poke_noanswer: Peer xxx is now unreachable

2006-07-11 Thread broadbandvoice


I was getting this message throughout yesterday in repitition, anyone experienced this before and what is the best solution?

Jul 10 12:19:03 NOTICE[13020]: chan_sip.c:11364 sip_poke_noanswer: Peer '4001' is now UNREACHABLE! Last qualify: 124Jul 10 12:19:03 NOTICE[13020]: chan_sip.c:11364 sip_poke_noanswer: Peer '4002' is now UNREACHABLE! Last qualify: 120Jul 10 12:19:05 NOTICE[13020]: chan_sip.c:11364 sip_poke_noanswer: Peer '4003' is now UNREACHABLE! Last qualify: 117Jul 10 12:19:07 NOTICE[13020]: chan_sip.c:11364 sip_poke_noanswer: Peer '4004' is now UNREACHABLE! Last qualify: 116Jul 10 12:19:08 NOTICE[13020]: chan_sip.c:11364 sip_poke_noanswer: Peer '4005' is now UNREACHABLE! Last qualify: 124Jul 10 12:19:08 NOTICE[13020]: chan_sip.c:11364 sip_poke_noanswer: Peer '4006' is now UNREACHABLE! Last qualify: 131Jul 10 12:19:08 NOTICE[13020]: chan_sip.c:11364 sip_poke_noanswer: Peer '4006' is now UNREACHABLE! Last qualify: 130Jul 10 12:19:10 NOTICE[13020]: chan_sip.c:9700 handle_response_peerpoke: Peer '4007' is now TOO LAGGED!
  (2142
ms / 2000ms)Jul 10 12:19:13 NOTICE[13020]: chan_sip.c:9694 handle_response_peerpoke: Peer '4001' is now REACHABLE! (112ms / 2000ms)Jul 10 12:19:13 NOTICE[13020]: chan_sip.c:9694 handle_response_peerpoke: Peer '4002' is now REACHABLE! (120ms / 2000ms)Jul 10 12:19:15 NOTICE[13020]: chan_sip.c:9694 handle_response_peerpoke: Peer '4003' is now REACHABLE! (118ms / 2000ms)Jul 10 12:19:18 NOTICE[13020]: chan_sip.c:9694 handle_response_peerpoke: Peer '4004' is now REACHABLE! (118ms / 2000ms)Jul 10 12:19:18 NOTICE[13020]: chan_sip.c:9694 handle_response_peerpoke: Peer '4005' is now REACHABLE! (114ms / 2000ms)Jul 10 12:19:18 NOTICE[13020]: chan_sip.c:9694 handle_response_peerpoke: Peer '4006' is now REACHABLE! (139ms / 2000ms)Jul 10 12:19:18 NOTICE[13020]: chan_sip.c:9694 handle_response_peerpoke: Peer '4007' is now REACHABLE! (136ms / 2000ms)Jul 10 12:19:20 NOTICE[13020]: chan_sip.c:9694 handle_response_peerpoke: Peer '4008' is now REACHABLE! (142ms / 2
 000ms)


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Re: [asterisk-users] DTMF

2006-07-06 Thread broadbandvoice



try setting your dial plan in sip.conf using dtmf = rfc2833
-- Original message -- From: El Flynn [EMAIL PROTECTED]  Rizwan Hisham wrote:   Hi,   i need to set the dtmf mode on my quintum tenor a400 gateway.   You might want to check the a400 manual on how to do that.i cant dial   any extension thru my normal digital phone which is connected to asterisk   thru the quintum gateway. it always falls to 's' extension. So plz help This is most likely a misconfiguration of your dialplan and/or sip.conf files.  it would help if you post it here?   Flynn___  --Bandwidth and Colocation provided by Easynews.com --   asterisk-users mailing list  To UNSUBSCRIBE or
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Re: [asterisk-users] Phones cutting out.....again - PLEASE HELP!!!

2006-07-06 Thread broadbandvoice

Do you have tetheral network analyser installed on server, that can be a good start, look at the analyses of the graphs. Also try pinging the CPE's and see if there is any latency. Do you also have the abilty to check the upstreams signals?

-- Original message -- From: "whois wes" [EMAIL PROTECTED] Hate to drag this one back up, butit's happening again. Overview of architecture: Dell poweredge 2850, running fedora core 4, asterisk 1.2.7.1, zaptel 1.2.5, and sangoma wanpipe 2.3.4 drivers. T1 interface card is the sangoma a104d with onboard echo can. Server is located in our data center and connected directly to our cisco 6513 core switch, so we have almost zero latency. The office having the issues is located several miles away and is connected via a 10Mbit fiber pipe, also low latency. Ping times between remote office and here are well under 10ms. T1's are robbed-bit, EM wink signalling --- (this may be cause, but want your input).
&
gt; Server load is averaging around 20%, plenty of memory, disk space, and bandwidth available. No QOS running on network. ulaw is the primary codec. Server is stable, and there are no extraneous services running, save mysql and httpd. Even running a processor intensive query doesn't trigger the droputs, they happen randomly. Phones are a mix of Eyebeam 1.5.5 and Eyebeam 1.10 3010n. Both types of phones are experiencing cutting out of the signal, mainly in the Rx stream, but occassional in the Tx stream as well. The cutting out was NOT occurring last night, and the phone server is being rebooted nightly. Nothing has changed AT ALL, and the problem has started occurring again. If I don't do ANYTHING at all today, there is a 50% chance that this will NOT occur tomorrow. In other words, SOMETHING is causing our phones to drop out, but whatever changes I m
ake se
em to have no effect. The problem will start and stop seeminly at it's own whim. --- Things I have tried: 1. changed from ulaw to gsm as primary codec - no change 2. disabled hardware echo can on A104D - no change 3. moved from asterisk 1.2.4 to 1.2.7.1, recompiled both several times - no change 4. have played with gain settings a bit, doesn't seem to make much difference --- At this point, i am nearing the end of my rope - i have rebuilt this machine three times now, and have recompiled the system at least a dozen times. We have gone from Digium hardware to Sangoma harware and back again. I have changed every conceivable setting on the phones to no avail. The problem will randomly disappear, only to come back a few days later. I can make a change, it seems to have an effect, then we're back to the
same o
ld thing again. I am in dire need of ANY help anyone can offer, this has been going on in some form for almost three months. Thanks for reading, Wes ___ --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

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RE: [asterisk-users] for you guys setting up customer offices...

2006-07-06 Thread broadbandvoice


I could have told you that. Ihave 4 handy tones wasting in my basement.

-- Original message -- From: "calvis" [EMAIL PROTECTED]  Polycom 501   Grandstreams are junk. (I have had bad experiences with them) -Original Message-  From: [EMAIL PROTECTED]  [mailto:[EMAIL PROTECTED] On Behalf Of Shaun  Sent: Thursday, July 06, 2006 12:45 PM  To: asterisk-users@lists.digium.com  Subject: [asterisk-users] for you guys setting up customer offices...   What brand/model phones are you using.   --   ~Shaun ___  --Bandwidth and Colocation provided by Easynews.com --   asterisk-users mailing list  To UNSUBSCRIBE or update opt
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Re: [Asterisk-Users] Now that Nufone is dead...

2006-07-04 Thread broadbandvoice


Try Termilink. www.termilink.net

-- Original message -- From: "Carlos Chavez" [EMAIL PROTECTED]  Now that Nufone is dead, what are other providers of 800 numbers that  work with Asterisk?   --  Carlos Chavez  Director de Tecnología  Telecomunicaciones Abiertas de México S.A. de C.V.  Tel: +52-55-91169161 Ext 2001   ___  --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 

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Re: [Asterisk-Users] Best VoIP provider for Asterisk

2006-07-04 Thread broadbandvoice


Termilink, at www.termilink.net

-- Original message -- From: "C F" [EMAIL PROTECTED]  Define best.   On 5/23/06, Crazy Boy <[EMAIL PROTECTED]>wrote:   Hi Friends, Can you please tell me who is the best VoIP Service Provider using Asterisk   (With trail version for sometime) . Waiting for your quick response. Thank   you. Regards,   Chandra.   __   Do You Yahoo!?   Tired of spam? Yahoo! Mail has the best spam protection around   http://mail.yahoo.com   ___   --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing l
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Re: SV: [Asterisk-Users] Running 40 active calls (too much f�r CPU?)

2006-07-04 Thread broadbandvoice

Are the phones behind a NAT? What is the processory memory size? Are the E1 channelized?

-- Original message -- From: [EMAIL PROTECTED] I should add that thease 25 calls where SIP (internal) to Zap (PSTN) calls. Mvh, Jan -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 4 juli 2006 09:41 Till: asterisk-users@lists.digium.com Ämne: [Asterisk-Users] Running 40 active calls (too much för CPU?) Hi, We're running asterisk 1.2.1 on a Dell PowerEdge 600SC (2.4 ghz) server connected to the PSTN through two E1 pipes to a TE405P. This has been running just fine for several months... But yesturday we connected a large number of softphone SIP clients (50) and 25 <
BR>
; of these where running simultaneous active calls on the INTERNAL ethernet using g711 (ulaw). We noticed that the sound was jagged just as if the CPU couldn't handle 25 calls (?!). I checked the CPU load and it never went over 55 % and memusage was low too. Does anyone know what could be the problem? Are there some kind of CPU spikes that make these cuts in the audio? If so, why on earth can't a 2,4 ghz processor handle 25 low-quality audio "tracks" on asterisk when I can run +50 cd-quality audio tracks when producing music? ANY help and/or comments would be appreciated since this is quite an acute problem. Regards, Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: h
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RE: [Asterisk-Users] Volume configuration on Polycom Soundpoint 501phone

2006-05-25 Thread broadbandvoice


Could not find your post for 4 months ago.

-- Original message -- From: "Anton Krall" [EMAIL PROTECTED]  Yes, check a post that I made about 4 months ago, I posted the cofig for  setting the speaker, handset and ring volumes ..   |-Original Message-  |From: [EMAIL PROTECTED]  |[mailto:[EMAIL PROTECTED] On Behalf Of  |Jerry Jones  |Sent: Thursday, May 04, 2006 3:15 PM  |To: Asterisk Users Mailing List - Non-Commercial Discussion  |Subject: Re: [Asterisk-Users] Volume configuration on Polycom  |Soundpoint 501phone  |  |Edit your config files to enable persistance  |  |Will remain across multiple calls, but not reboots  |  |  |On May 4, 2006, at 2:51 PM, Jim Freeze wrote:  | 
  |
 We are using the polycom 501 phones, and are having some challenges  | with the volume setting. When a phone call comes in, the  |user ups the  | volume for the handset, but they have to repeat that for every call.  |  | Currently, the volume level seems to reset itself at about 60%.  | Is there a way for the user to change their default volume level?  |  | Thanks  |  | --  | Jim Freeze  | ___  | --Bandwidth and Colocation provided by Easynews.com --  |  | Asterisk-Users mailing list  | To UNSUBSCRIBE or update options visit:  | http://lists.digium.com/mailman/listinfo/asterisk-users  |  |___  |--Bandwidth and Colocation provided by Easynews.com --  |  
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[Asterisk-Users] PANASONIC KX-TS208W - Speakerphone Incompatible With Asterisk 1.2.3


I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does not work with it. It works fine when you pick up the handset. Anyone experinced this problem before, the speaker works fine with Verizon line. The phone is behind a Linsys router RT31P2.

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RE: [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone IncompatibleWith Asterisk 1.2.3



You got me there, it's at a customer's premise. I will have to find out from them, if it a single pair.
-- Original message -- From: "Steve Totaro" [EMAIL PROTECTED]  I am not familiar with that phone. Is it single pair?-Original Message-  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]  Sent: Sat 4/22/2006 12:13 PM  To: asterisk-users@lists.digium.com  Cc:  Subject: [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone  IncompatibleWith Asterisk 1.2.3   I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does not work  with it. It works fine when you pick up the handset. Anyone experinced this  problem before, the speaker works fine with Verizon line. The phone is behind a  Linsys router RT31P2.

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Re: [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone Incompatible With Asterisk 1.2.3


Thanks for the response, I'll ask the client to change batteries, though it is a new phone less than two weeks. is there any reason why the Lanline(Verizon) work and not the Asterisk? The only differences is the Asterisk, Linksys router and the DSL modem. One of these 3 should be interfering.

-- Original message -- From: John Novack [EMAIL PROTECTED]  [EMAIL PROTECTED] wrote:I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does   not work with it. It works fine when you pick up the handset. Anyone   experinced this problem before, the speaker works fine with Verizon   line. The phone is behind a Linsys router RT31P2.   Replace the batteries! Alkaline only, replace every 6 months   1.2.3 is also defective for other reasons. Upgrade   Using a TDM400, an ATA or ??   The phone works best with a 48V 20mA or better loop, so the FXS source   voltage may have an effect, and this cheap phone has no previsions   for external power. 
   
  John Novack___  --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 

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RE: [Asterisk-Users] Don't see my post



Gafachi can, I've been using them with you problems.
-- Original message -- From: [EMAIL PROTECTED] 








First of all, try sending it to the asterisk-biz list.





From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John RichSent: Monday, April 17, 2006 10:53 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Don't see my post

Hi Folks,I have posted a couple of message to the list and do see them, even after waitin for long time (2 days). Can someone please point me to the rules for posting to this list? I think it had to do with the subject that I was looking for. I was looking for IAX terminiation service that can handle high volumes.ThanksJohn.



Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.

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RE: [Asterisk-Users] How to restrict simultaneous phone registrations


disable three-way calling, restric channels to one per call.

-- Original message -- From: "Bill Gibbs" [EMAIL PROTECTED]  I say just bill the user at extension 333 it's his responsibility to  keep the login info private. If he disputes it, refund the first time  then change the password to something really complicated then start  billing him if it keeps happening after that!   Bill   -Original Message-  From: [EMAIL PROTECTED]  [mailto:[EMAIL PROTECTED] On Behalf Of Bryan  Mahin  Sent: Wednesday, April 05, 2006 10:50 PM  To: asterisk-users@lists.digium.com  Subject: RE: [Asterisk-Users] How to restrict simultaneous phone  registrations   :) I should rephrase my question. And included a bit more information on  wha
 t I am
 trying to accomplish.   Solution 1 (preferred)   I am working on an asterisk installation where most end users will use  softphones. If I am not able to lock down calling to one phone at a  time, the end users will share their login information with friends,  family, neighbors, and the some girl they meet on myspace.   Currently, I am able to register two phones at separate locations with  the same account on each phone and make concurrent calls.   For example, If I login extension 333 at location A, and 333 at location  B, simultaneous calls can be placed from both phones at the exact same  time. I only want calls placed from extension 333 to work from either A  or B not A and B concurrently.   Here is my ideal solution. Location A wants to make a call, but location  B has a call in progress. Location B has to either close their pho
 ne, or
  hang up before Location A can make the call.OR.. Solution 2. :)  A way I can distinguish in my CDR the IP address or some other  recognizable difference between the two locations when they make  concurrent calls using the same extension. The complication here is; I  can currently the log IP addresses, but as the end phones are on the  internet, Nat'd, and I am using a siparator for traversal. As a result,  my logs show the IP address of the siparator and I don't have any other  data to distinguish the end phones.   OR.. Solution 2.5  One thought I've had is to send logs from the siparator to a syslog  server, parse them, hunt for simultaneous calls placed by the same  accounts from different locations, and bill the end users accordingly.  But I really dislike this idea as no one likes to be hit with  surcharges.  
 ; Any 
help or insight is greatly appreciated.   Thanks again,  Bryan Mahin-Original Message-  From: [EMAIL PROTECTED]  [mailto:[EMAIL PROTECTED] On Behalf Of Eric  "ManxPower" Wieling  Sent: Wednesday, April 05, 2006 7:50 PM  To: Asterisk Users Mailing List - Non-Commercial Discussion  Subject: Re: [Asterisk-Users] How to restrict simultaneous phone  registrations   Bryan Mahin wrote:   Hello all, I am looking for a way to restrict users from logging in two separate   phones with the same authorization name/password at the same time.   Meaning, I only want users to be able to place a call from one phone  in   one location, but have the ability to move from computer to computer.   Has anyone found any sort of solut
 ion fo
r this type scenario?   This is a non-issue, because a second registration to the same account  will override and previous registrations for that account.  ___  --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   Please visit us @ www.uneta.com   ___  --Bandwidth and Colocation provided by Easynews.com --   Asterisk-Users mailing list  To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users  ___  --Bandwidth and Colocation provided by Easynews.com --   Asterisk-Users mailing list  To UNSUBSCRIBE
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RE: [Asterisk-Users] No DTMF


I had the same problem, I reloaded Asterisk 1.2.3 and set the dtmf 2833 that fixed it.

-- Original message -- From: "Mark Edwards" [EMAIL PROTECTED] 




Try dtmfmode=info and see if that works.

Mark

-Original Message-From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Thursday, 9 March 2006 6:08 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] No DTMF


Some one was on my server making changes to my sip.conf files. I am now having trouble with DTMF. No matter what I use (inband,auto,rfc2833) the dtmf tones seem to not come thru. I compared it to the wiki and all the configs seem to be in order.



Here is my sip.conf



[general]disallow=all;allow=g729 ; requires license for g729allow=ulawport = 5060nat=yescontext=from-sipbindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)maxexpirey=4800 ; Maximum expiration for registrationsdefaultexpirey=1800 ; Default expiration for registrationscanreinvite=no ; Allow clients to directly connect if set to yes. Set to no if behind NAT.tos=reliabilitysrvlookup=yes ; Enable DNS SRV lookups on outbound callsvideosupport=no ; Turn on support for SIP videodtmfmode=rfc2833 ;rfc2833 ;inband ;rfc2833 ; DTMF inband need to be set here.pedantic=noexternip=..XXX

;Sip Mediaregister = XX:[EMAIL PROTECTED]/7322761368

[sipmedia6]type=frienduser=XX ;(Phone Number)username=XX ;(Phone Number)fromuser=XX ;(Phone Number)authname=XX ;(Phone Number)secret= ;(SIP Password)host=sip.sipmedia.com disallow=allallow=ulawcontext=ServerHighwayrealm=sip1.xchangetele.comfromdomain=sip.sipmedia.comdtmfmode=rfc2833canreinvite=no insecure=very



Here is my extensions.conf

[general]static=yeswriteprotect=yes

[ServerHighway];Play Server Highway IVR

Exten = s,1,Background(server-highway-ivr)Exten = s,2,Background(blank-file-10)

Exten = 1,1,Ringing()Exten = 1,2,Wait(15)Exten = 1,3,Macro(stdexten,9511,9511)Exten = 2,1,Ringing()Exten = 2,2,Wait(15)Exten = 2,3,Macro(stdexten,9512,9512)Exten = 3,1,Ringing()Exten = 3,2,Wait(15)Exten = 3,3,Macro(stdexten,9513,9513)Exten = 4,1,Ringing()Exten = 4,2,Wait(15)Exten = 4,3,Macro(stdexten,9514,9514)Exten = i,1,Background(invalid)Exten = i,2,Goto(s,1)

Exten = t,1,Goto(s,1)

exten = 9,1,Goto(s,1);Extension To Record Main IVR Messageexten = 500,1,Authenticate(XXX)exten = 500,2,Record(ServerHighwayIvr:gsm)



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Re: [Asterisk-Users] Redirecting to another service/server

2006-04-21 Thread broadbandvoice


Did you get an answer to this? I am interested in SIP to SIP calls on other networks thereby by-passing the pstn.

-- Original message -- From: Nick Hoffman [EMAIL PROTECTED]  Hi guys. Without having a FWD account, can Asterisk redirect calls to FWD?   For instance, an extension behind Asterisk dials 99751234, and Asterisk  says "that starts with 99. let's strip off the 99 and call 751234 at FWD,  IE: sip:[EMAIL PROTECTED]:5060".   Is that possible, or would services such as FWD reject the call since the  device making the call (Asterisk) hasn't registered?   Thanks!  -- Nick  e: [EMAIL PROTECTED]  p: +61 7 5591 3588  f: +61 7 5591 6588   If you receive this email by mistake, please notify us and do not make any  use of the email. We do not waive any privilege, confidentiality or  copyright associated wi
 th it.
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Re: [Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud



I'm running 1.2.3 and that seems to be the most stable version, had problems with other versions too.
-- Original message -- From: "Brian Roy" [EMAIL PROTECTED] 
On 3/3/06, Gary Richardson [EMAIL PROTECTED] wrote: 

I'm running 1.2.4 and just about every call is cut short. I'm using Cisco IP phones as end points. All the outbound calls are routed via SIP through a PRI line attached to a Cisco 2811..


I'm running 1.2.1 and most of mine get cut short too. I posted this on the list a few months ago and nobody had any suggestions. BJ said I should probably post a bug on it but I haven't had time to continue to troubleshoot it. I will go to 1.2.4 (now 5 probably) and see if mine goes away. I've been watching change logs and hadn't seen anything surrounding mixmonitor so I've let it go.

Please continue to update us if anyone gets some resolution. I'm glad to know there are lots of us experiencing this. That should be the catalyst to get it fixed.

-Brian



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RE: [Asterisk-Users] incoming calls dropout on PRI over TE110p


Try 1.2.3, it works fine.

-- Original message -- From: "James Sturges" [EMAIL PROTECTED]  I would not upgrade to 1.2.x yet, I did and now have taken asterisk out of  the site. It is sending CRC errors )to Telsta, drops all calls once a day  for 1 second, calls getting stuck, quite unpleasant!   I was advised to roll back to 1.0.9 Asterisk, Zaptel and Libpri.   James-Original Message-  From: [EMAIL PROTECTED]  [mailto:[EMAIL PROTECTED] On Behalf Of Paul C  Sent: Wednesday, 1 March 2006 4:15 PM  To: Asterisk Users Mailing List - Non-Commercial Discussion  Subject: Re: [Asterisk-Users] incoming calls dropout on PRI over TE110pPaul C wrote:   I am running Asterisk 
 1.0.9 
and have been running all my calls through a   VSP over a IAX2 trunk however we have recently purchased and connected a   TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can makeoutgoing calls via it fine, however incoming calls are dropped after a   few seconds ( or as soon as a command like Playback, or the call is   picked up if forwarded to a SIP extensions ).SNIP   overlapdial should usually be no in my experience.Okay I've turned that to no with no change. I've just got off the phone to  Optus and apparently they had a client in melbourne last week and they fixed   the problem by turning crc checking off at the optus end. I don't suppose  that was anybody on here ?   ___ &
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Re: [Asterisk-Users] Dialplan - strip IDD prefix and insert another


How's tou're service with Sellvoip, I was not able to intergrate them into my system and they had no phone support. I'm using Gafachi now but prefer the rates Sellvoip provide.

-- Original message -- From: "AR Tarzi" [EMAIL PROTECTED] 



SellVoIP appears to follow a US dialplan. A US numberis dialled as 1NXXNXX whereas an international (to the US) numberis dialled as 011X.
Frankly, I didn't ask whether international numbers like Barbados where the code remains as 1 butare international (to the US) need the 011 or can be dialled directly but that's not really my concern. I've assumed they don't.

Most of the world uses 00 as the internation prefix code, therefore I have to ask:

Howcan I "strip" the 00 and insert 011 in one entry in the dialplan. I'm stripping the 00 and passing the rest of the numbers for numbersdialled as001X. (as in: 00|1XX.) but in case of numbers out of the US, how would I insert the 011 ?



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X-WAB-GENDER:2
URL;WORK:www.arabbanking.com
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EMAIL;INTERNET:[EMAIL PROTECTED]
EMAIL;INTERNET:[EMAIL PROTECTED]
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EMAIL;INTERNET:[EMAIL PROTECTED]
REV:20060305T155750Z
END:VCARD

RE: [Asterisk-Users] re: Sixtel Services


there 2 types of inbound metered and unmetered. unmetered is unlimited inbound and metered charges per the minutes.

-- Original message -- From: "Steve Totaro" [EMAIL PROTECTED]  Inbound should be free as far as I am concerned unless you have a toll  free number. Thanks,  Steve Totaro _   From: VIC IP Communications [mailto:[EMAIL PROTECTED]  Sent: Sunday, March 05, 2006 11:28 AM  To: Asterisk-Users@lists.digium.com  Subject: [Asterisk-Users] re: Sixtel Services Hi,   Companies like DIDx and Sixtel, when they state DIDs at $XX.XX per month  and $XX.XX per minute/monthly,  do these companies provide inbound and outbound routing of calls, or are  these rates strictly for inbound   Call routing of DIDs?  
  &
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Re: [Asterisk-Users] re: Sixtel Services


They're for inbound only though some of them provide termination services

-- Original message -- From: "VIC IP Communications" [EMAIL PROTECTED] 






Hi,Companies like DIDx and Sixtel, when they state DIDs at $XX.XX per month and $XX.XX per minute/monthly,do these companies provide inbound and outbound routing of calls, or are these rates strictly for inbound
Call routing of DIDs?



Thanks.

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Re: [Asterisk-Users] RE: Asterisk-Users Digest, Vol 20, Issue 31