Re: [asterisk-users] How set debug file for RxFax application
How is your system configured?
Debug output of faild faxes?
This kind of information is needed to help you!
Regards,
Juan
khalid touati wrote:
can anyone help me out in this, a big number of my faxes
are lost everyday! i would really appreciate any help on how i can
tweak asterisk (rxfax) to receive all faxes!
2010/4/2 khalid touati khalidtou...@gmail.com
i
went ahead and i used this line:exten =
3772,n,rxfax(${FAXFILE}|debug) as it says in the rxfax tutorial (also
because i'm not sure that FAXOPT is supported by asterisk 1.2), but no
output neither in the CLI or in a file. so is there any body who knows
about that?
i'll appreciate any help!
2010/4/2 khalid touati khalidtou...@gmail.com
thank
you guys for responses,
Danny-, am i going to receive debug info in the CLI or in a default
file (/var/log /*)? is FAXOPT supported whitin asterisk 1.2 (sorry I
forgot to mention that i am using 1.2)?
2010/4/2 Tzafrir Cohen tzafrir.co...@xorcom.com
On Fri, Apr 02, 2010 at 10:11:59AM -0400, khalid touati
wrote:
Hi Guys,
do any body know how to receive debug info on RxFAX application? i
am
experiencing a lot of fax failures and can't guess the reason
behind.
Thank you very much for any help!
They have a hard-wired log file. Make sure Asterisk can
write to it.
--
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Re: [asterisk-users] RPID on called party
Try using sendrpid=yes on sip.conf Regards, Juan Ondrej Valousek wrote: Hello, Did anyone manage to force asterisk to put Remote-party-ID attribute on the SIP outgoing call? I.e. When A calls B, I want that A gets a name of B displayed on his phone. Note that name of A gets displayed on the B's phone fine, but this is not what I want. This works with Cisco Call manager fine - the RPID is sent as a part of the response to the SIP INVITE this way: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.60.20:5060;branch=z9hG4bK42892c32;rport From: Ondrej Valousek sip:7...@192.168.60.20 sip:7...@192.168.60.20 ;tag=as4786d518 To: sip:1...@192.168.62.12 sip:1...@192.168.62.12 ;tag=f75ff5d8-1023-4240-bc4b-d7eeb6d0d77d-42063104 Date: Tue, 30 Mar 2010 13:53:15 GMT Call-ID: 465a9c200587260d164f451409489...@192.168.60.20 CSeq: 102 INVITE Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY Allow-Events: presence *Remote-Party-ID: Paul Ryan sip:1...@192.168.62.12 sip:1...@192.168.62.12 ;party=called;screen=yes;privacy=off* Contact: sip:1...@192.168.62.12:5060 sip:1...@192.168.62.12:5060 Content-Length: 0 But I can not make it working with Asterisk. Does anyone have any glue how to achieve this WITHOUT patching asterisk? I am happy to upgrade to the latest/greatest version, I just do not want to patch. Many thanks, Ondrej -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Connection Question
Depends on the configuration you make. For example, if you want to route the call giving the Mitel a new desrination or prefix, you can use Transfer dialplan app. Transfer before answering the call will be redirected with SIP 302. If the call is to be anwered on *, then canreinvite set to yes or directrtp set to yes can help you. Saludos, Juan E. Rodríguez -Original Message- From: Kenneth Noisewater noisewater...@gmail.com Date: Thu, 1 Apr 2010 16:50:47 To: asterisk-users@lists.digium.com Subject: [asterisk-users] SIP Connection Question -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Connection Question
If * answers the call, it will be on the loop but with canreinvite or directrtp the media can be out of * and redirected to the final end point even if signaling goes through *. For the trunk, you can have multiple simultaneous calls. I do not know about Mitel's licensing but with only one trunk you can have as much calls as * supports. Saludos, Juan E. Rodríguez -Original Message- From: Dr. Kenneth Noisewater noisewater...@gmail.com Date: Thu, 01 Apr 2010 19:35:54 To: jerdg...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP Connection Question On 4/1/2010 5:09 PM, Juan E. Rodríguez wrote: Depends on the configuration you make. For example, if you want to route the call giving the Mitel a new desrination or prefix, you can use Transfer dialplan app. Transfer before answering the call will be redirected with SIP 302. If the call is to be anwered on *, then canreinvite set to yes or directrtp set to yes can help you. Saludos, Juan E. Rodríguez -Original Message- From: Kenneth Noisewaternoisewater...@gmail.com Date: Thu, 1 Apr 2010 16:50:47 To:asterisk-users@lists.digium.com Subject: [asterisk-users] SIP Connection Question OK, so for instance if I passed a call to Asterisk and grabbed CID info and did some lookups and then transferred it back to mitel to route to a user, then * would be out of the call path (loop, whatever). But, if I were to answer that call in * with an IVR to collect caller input to use and then transferred the call back to the Mitel to route to the endpoint, * would remain in the call. Is that a correct understanding? Also one more question, and please excuse my ignorance (I'm just a developer with pretty limited knowledge on the telephony side of things): When I talk about connecting the Mitel box and the Asterisk box together via a SIP trunk, is that trunk equal to 1 analog line, or channel or whatever, or can I make as many connections as I want on that trunk? Again, my knowledge is a bit limited, and thusfar people have been using a lot of terms interchangably with me to add to my confusion :). This only concerns me because I'm pretty sure we have to buy a license for each SIP trunk with Mitel. It would be really great if I could work out a solution like this, it will allow me to prove Asterisk's worth to my management, and open up a lot of doors for us and our internal apps. The Mitel SDK is unfortunately rather limited, but management is not in any way interested in jumping ship from Mitel to Asterisk. Personally, I say jump, I've had great experience with Asterisk, even in fairly heavy use situations. Anyone have any input on selling Asterisk to higher up's? I know there is the whole enterprise support aspect, but my team manages the Mitel stuff as it is anyway, and I think we'd all much rather be dealing with Asterisk/SER as the core solution. Thanks everyone for your input! Kenny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confusion on call forwarding
You need promiscredir set to yes on sip.conf --Mensaje original-- De: Richard Kenner Remitente: asterisk-users-boun...@lists.digium.com Para: asterisk-users@lists.digium.com Responder a: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] Confusion on call forwarding Enviado: 30 Mar, 2010 13:38 I'm confused. What does Asterisk do when it gets a 302 with a new number to forward to? Is there anything I have to do in the dialplan to make this work? I can't find any clear documentation on this issue. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Saludos, Juan E. Rodríguez -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confusion on call forwarding
After setting promiscredir set to yes * is goign to send the call to the first desrination on the Contact header. --Mensaje original-- De: Richard Kenner Remitente: asterisk-users-boun...@lists.digium.com Para: asterisk-users@lists.digium.com Responder a: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Confusion on call forwarding Enviado: 30 Mar, 2010 14:56 You need promiscredir set to yes on sip.conf And then what do I do in the dialplan? I.e., what context is the redirect number interpreted in? Google searches on this issue show inconsistent and contradictory information. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Saludos, Juan E. Rodríguez -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing wont pass the number
When you say 'a2billing' won't pass the number, you mean you are calling to an IVR or something like that. And when did you dial you destination number twice??? Saludos, Juan E. Rodríguez -Original Message- From: Nathanial Allan nathanial.al...@gmail.com Date: Tue, 30 Mar 2010 13:08:24 To: asterisk-users@lists.digium.com Subject: [asterisk-users] a2billing wont pass the number I am running into an issue with A2Billing. I will explain first of all that everything else works! the system is 90% complete its just this one small problem I am running into. So my problem is that when I place a call, 1. I dial my number that I want and A2Billing gets activated 2. it asks for my pin, upon successful entry of my pin A2Billing then 3. prompts me for my phone number then 4. The call goes out (and actually connects for the record) So I am entering my destination phone number twice which is not the worst thing that can happen, though it is a little annoying Any light that you can shine on this problem would be greatly appreciated as I have been working on it for too long now and I want to get a product! Thank You NallaN -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What does this error message mean
Show sip.conf and extensions.conf related part. Maybe I misread but did you mention you have a exten... Line in sip.conf??? The error is because the received user is not the same as the configured one. --Mensaje original-- De: Ira Remitente: asterisk-users-boun...@lists.digium.com Para: Asterisk Users Mailing List - Non-Commercial Discussion Responder a: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] What does this error message mean Enviado: 26 Mar, 2010 21:01 At 05:47 PM 3/26/2010, you wrote: You have used this same username/password combination for another SIP client, or maybe the same one but with different IP. Even when that one is offline from some time on, Asterisk doesn't renew it's internal database, so still thinks it might be somewhere there. Why thanks. The sad part is it means madly flailing has exactly the opposite effect as you might expect. Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Saludos, Juan E. Rodríguez -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need help on setup rtp directly between 2 sipclients
Try setting canreinvite and nat to no for those extensions. Saludos, Juan E. Rodríguez -Original Message- From: Alyed al...@vivoxie.com Date: Fri, 26 Mar 2010 10:56:50 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] need help on setup rtp directly between 2 sip clients -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get Sip response codes in Dialplan?
Try using DIALSTATUS. --Mensaje original-- De: Zhang Shukun Remitente: asterisk-users-boun...@lists.digium.com Para: Asterisk Users Mailing List - Non-Commercial Discussion Responder a: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] How to get Sip response codes in Dialplan? Enviado: 24 Mar, 2010 23:29 hi ,all when a Dial or Queue excutes, a sip response code will return. like == Using SIP RTP CoS mark 5 -- Got SIP response 502 Bad Gateway back from 211.150.119.32 -- SIP/95040-004a is circuit-busy -- Nobody picked up in 2000 ms My quesion is how to get the response code in the dial plan immediatelly in order to do different thing according the returned codes? for example: a queue response code is busy now i will queue another number immediately not let the user waiting for the timeout. Thanks! -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Saludos, Juan E. Rodríguez -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hard Phone with SMS
Johann: Do you know how is the SMS sent over the IP, does it use SIP INFO message or somthing like that? Regards, Juan Johann Steinwendtner wrote: randulo schrieb: 2009/10/9 "Juan E. Rodrguez" jerdg...@gmail.com: Does any one know about a SIP hard phone capable of sending SMS messages (Or a SIP MESSAGE) that could be read from Asterisk dial plan?? The Gigaset S675IP series of DECT/SIP phone has SMS capability, but not sure it can work with Asteris. Yes, they do. (app_sms) Make sure you have installed the latest FW. Before, they sent the SMS out on the analog port only. Hans ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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] Faxing: Anyone have a compiled executable?
I have it running on * 1.4.19. You can get it on the internet, .so and an intaller that checks for dependencies. --Mensaje original-- De: Doug Remitente: asterisk-users-boun...@lists.digium.com Para: asterisk-users@lists.digium.com Responder a: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Faxing: Anyone have a compiled executable? Enviado: 7 Ene, 2010 01:41 At 16:49 1/5/2010, Tzafrir Cohen wrote: On Tue, Jan 05, 2010 at 04:24:37PM -0600, Doug wrote: Hi, Having problems with getting either RxFax or FaxReceive to compile. Running Asterisk 1.4 on CentOS 5. What version of SpanDSP do you use? spandsp-0.0.6pre12.tgz and: libtiff-3.8.2-7.el5_3.4 libtiff-devel-3.8.2-7.el5_3.4 Which do you recommend? ___ -- 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 Saludos, Juan E. Rodríguez ___ -- 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] SIP Issue
You should set the ddwhome variable with the Set function or declare it on the global context. Try the Dial command with the dial string directly, before using the variable. Fro debugging purposes you should set debug and verbose at least to 10 and check the logs. Regards, Juan James A. Shigley wrote: What do you mean I should use a global function. I'm kind both well versed and a newb to * James Shigley -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan E. Rodríguez Sent: Monday, December 28, 2009 12:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Issue Is ddwhome defined in global context?? If so, then you should use global function. Paste asterisk log to check. Saludos, Juan E. Rodríguez -Original Message- From: James A. Shigley j...@answeringserv.com Date: Mon, 28 Dec 2009 12:11:35 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [asterisk-users] SIP Issue ___ -- 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 ___ -- 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] Does A2Billing has mial list?
A2billing forum has a lot of information and questions are answered very fast. Try searching on the forum before posting, cause the answer may be there already. forum.asterisk2billing.org/ Regards, Juan Bruce Nik wrote: Hi Sucan, A2Billing doesn't have a mailing list but you may ask your specific question on A2billing Forum or maybe even here. This may be of intrest to you if you have an installation question: A2Billing automated install script : http://a2billing2asterisk.googlepages.com -Bruce On Tue, Dec 29, 2009 at 1:07 AM, Zhang Shukun bit...@gmail.com wrote: hi, Does A2Billing has mial list? -- Thanks, Sucan ___ -- 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 ___ -- 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] SIP Issue
Is ddwhome defined in global context?? If so, then you should use global function. Paste asterisk log to check. Saludos, Juan E. Rodríguez -Original Message- From: James A. Shigley j...@answeringserv.com Date: Mon, 28 Dec 2009 12:11:35 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [asterisk-users] SIP Issue ___ -- 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] pattern matching
You do not need to use pattern matching if you know the extension you are going to receive. Check the spelling on the dialplan if it does not work. You can start at the duplicated comma of the 34102. --Mensaje original-- De: Thomas Perron Remitente: asterisk-users-boun...@lists.digium.com Para: asterisk-users@lists.digium.com Responder a: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] pattern matching Enviado: 26 Dic, 2009 09:36 I want to ensure that only this extension is executed. But, I have others that are similar. I want: exten = 34101,1,Answer() exten = 34101,n,Record(34101:gsm) ; 34101 test zip code exten = 34101,n,Playback(34101) exten = 34101,n,Hangup Is this correct or do I need to have each of the four statements lead with an underscore (_) to make an exact match? Other code looks similar so I don't want the 102 to connect when I am dialing 101 exten = 34102,1,Answer() exten = 34102,,n,Record(34102:gsm) ; 34102 test zip code exten = 34102,n,Playback(34102) exten = 34102,n,Hangup ___ -- 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 Saludos, Juan E. Rodríguez ___ -- 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] sip realtime question
I am not sure, but I think you will get nothing with those commands if realtime
cathing is not set.
--Original Message--
From: Emre Kurnaz
Sender: asterisk-users-boun...@lists.digium.com
To: asterisk-users@lists.digium.com
ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] sip realtime question
Sent: Dec 11, 2009 4:02 AM
Hi everybody,
First of all i am sorry my English :)
i want to configure my asterisk server as a sip server that stores sip users in
the mysql database connecting directly over odbc driver. My odbc configuration
works as below
[r...@ao042 asterisk]# isql -v asterisk
+---+
| Connected!|
| |
| sql-statement |
| help [tablename] |
| quit |
| |
+---+
and i did refer to the site
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip . my res_odbc file as;
[r...@ao042 asterisk]# cat res_odbc.conf
[asterisk]
enabled = yes
dsn = asterisk
username = asterisk
password = ***
pre-connect = yes
[r...@ao042 asterisk]# grep -Ev '^;|^$' extconfig.conf
[settings]
sipusers = odbc,asterisk,sip_buddies
sippeers = odbc,asterisk,sip_buddies
and i created the asterisk database with sip_buddies table.
Here is my problem:
In asterisk console when i run the following command i get the answer,
ao042*CLI realtime load sipusers name 100
Column Name Column Value
id 1
name 100
host dynamic
nat no
type friend
cancallforward yes
canreinvite yes
secret Deneme01
disallow all
allow g729
allow ilbc
allow gsm
allow ulaw
allow alaw
port 5060
regseconds 0
lastms 0
username 100
but the following commands returns nothing
ao042*CLI sip show users
Username Secret Accountcode Def.Context
ACL NAT
ao042*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
Realtime
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]
besides it does not query anything. So what am i missing? Is there anything
that i should mentioned in the sip.conf file?
by the way i am using RHEL 5.4 with 2.6.18-164.el5 kernel -
asterisk16-1.6.0.17-1_centos5 rpm
Any help would be appreciated...
--
Emre Kurnaz
ITU/BIDB | Istanbul Technical University /
Information Technologies Office
Sistem Destek Grubu| System Support Team
RHCE : 805009174841679
Yar¹ Zamanl¹ Ö»renci Koordinatörü | Part-Time Student Manager
kurn...@itu.edu.tr
+90 0212 285 3930
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Saludos,
Juan E. Rodríguez
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Re: [asterisk-users] Can't restart asterisk from script
You should replace the single quote with double quote. --Original Message-- From: Michelle Dupuis Sender: asterisk-users-boun...@lists.digium.com To: 'Asterisk Users List' ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't restart asterisk from script Sent: Dec 9, 2009 10:59 PM But the error message in my log shows the error to be from asterisk, so I'm guessing I'm sending a parameter incorrectly to asterisk - which fits with the no quote theory -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bill Kenworthy Sent: Wednesday, December 09, 2009 9:31 PM To: Asterisk Users List Subject: Re: [asterisk-users] Can't restart asterisk from script Keep in mond that cron usually has a very abbreviated environment for security reasons - you may need to set the PATH or other environment variables in the crontab to get it to work. Billk On Wed, 2009-12-09 at 20:55 -0500, Michelle Dupuis wrote: Interesting...I'll try that. Thanks __ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Wednesday, December 09, 2009 8:47 PM To: Asterisk Users List Subject: Re: [asterisk-users] Can't restart asterisk from script Doug Lytle wrote: Warren Selby wrote: On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca mailto:supp...@ocg.ca wrote: I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x restart gracefully I have the following cron job: /usr/sbin/asterisk -r -x 'restart when convenient' Doug You probably don't need the single or double quotes at all. I have never used any quoting in crontab. Lyle Giese LCR Computer Services, 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 ___ -- 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 Saludos, Juan E. Rodríguez ___ -- 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 SS7 Sigtran Protocol
Right now, I think it does not. Look out for it at: asterisk-...@lists.digium.com Regards, Juan Khaled W Chehab wrote: Dears, Do Asterisk support SS7 SIGTRAN(SS7 over IP) protocol ? And how to integrate Regards Khaled Chehab NGN Eng. Operations Office - Lebanon Office: +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.Xplorium.com * NoemployeeoragentisauthorizedtoconcludeanybindingagreementonbehalfofXploriumwithanotherpartybye-mailwithoutexpresswrittenconfirmationbyanofficerofXplorium.AnyviewsexpressedbyanindividualinthiselectronicmessagedonotnecessarilyreflectviewsofXploriumoritssubsidiariesandassociates. Thiselectronicmessageanditsattachmentsaresolelyaddressedtotheaddressee(s),andcontainconfidentialinformationprotectedfromdisclosurebelongingtoXplorium. Ifyouarenottheintendedaddresseeofthiselectronicmessageanditsattachments,kindlydeleteitimmediatelyfromyoursystemandnotifythesenderbyelectronicmail.Youmustnotcopythismessageorattachmentordiscloseitscontenttoanyotherperson. Xploriumdoesnotguaranteetheintegrityofthiselectronicmessageandanyofitsattachments,orthattheyarefreefromcomputervirusesorotherdefects. * ___ -- 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 302 Moved Temporarily
Hello, I have an * installation that sometimes receives a 302 "Moved Temporarily" response to an INVITE. * sends the ACK but takes about 30 seconds to start the new INVITE to the new destination (from Contact Header). I have set core debugging to 20 but do not see any abnormal message. [provider] type=friend host=a.reachable.host.ip context=incoming_context dtmfmode=rfc2833 canreinvite=yes qualify=yes The call is generated from a PHP AGI script with the Dial "RrCL" options. Does any one have an idea why it could be lasting about 30 seconds to start the new call?? Regards, Juan ___ -- 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] IVR
As I see here, you do not have to include the big10 context inside the default context, as you have an extension defined to reach that context and its extention is start extension. If the cleveland-IVR is based on the start extension too, the same applies. Besides that, it would work...(maybe not the way you expect... :-) ) Regards, Juan Thomas Perron wrote: Is this going to work: [default] include = stdexten include = big10-IVR include = cleveland-IVR exten = _17035745353,1,Goto(big10-IVR,s,1) exten = _15672528431,1,Goto(cleveland-IVR,s,1) [big10-IVR] exten = s,1,Answer() exten = s,n,Background(dir-welcome) ;exten = s,n,WaitExten(1) ;exten = s,n,Background(astcc-please-enter-your) ___ -- 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] Unable to set TOS to 184?
If you already mangle packets with IPTABLES, then you should comment the line[s] tos_* on sip.conf. Regards, Juan Bart Fisher wrote: I don't understand this message: [2009-10-29 16:31:51] WARNING[28510]: rtp.c:1997 ast_rtp_settos: Unable to set TOS to 184 From what I have read the reason is asterisk can't set TOS if not running in root. Mine is running as asterisk. I found one post that says to run at boot: #!/bin/bash /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 4569 -j DSCP --set-dscp-class ef /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 1:2 -j DSCP --set-dscp-class ef /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 5060 -j DSCP --set-dscp-class ef Does this make sense? Is this the only method to end ths warning? Thanks, Bart ___ -- 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 302 Moved Temporarily
Hello, I have an * installation that sometimes receives a 302 "Moved Temporarily" response to an INVITE. * sends the ACK but takes about 30 seconds to start the new INVITE to the new destination (from Contact Header). I have set core debugging to 20 but do not see any abnormal message. [provider] type=friend host=a.reachable.host.ip context=incoming_context dtmfmode=rfc2833 canreinvite=yes qualify=yes The call is generated from a PHP AGI script with the Dial "RrCL" options. Does any one have an idea why it could be lasting about 30 seconds to start the new call?? Regards, Juan ___ -- 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] Dynamic DNS trunk
If the trunk is a dynamic IP you need the other end to register to Asterisk, so letting Asterisk know the new IP. Regards, Juan B.Masoud @ SH wrote: I have a trunk, and its host=dynamic dns. The problem is, when the IP changes the Sip show peers Still show the old IP of the DNS, I have to reload and save the configuration again so that asterisk recognize the new IP of the DNS. Any idea how to automate such a thing? Or how can I keep asterisk to deal with NAMES as NAMES, and IPs as IPs. Let me know. Thanks. ___ -- 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] Best Firewall Suggestions?
Check out ClarkConnect and SmoothWall. Regards, Juan Steve Totaro wrote: On Tue, Oct 13, 2009 at 2:41 PM, SIP s...@arcdiv.com wrote: David Wathen wrote: Hi, My customer has a outdated firewall that is also presenting a NAT nightmare for getting the Asterisk server reachable from the internet. What firewalls work good with VOIP? I really want to steer away from any ALG supported firewall. I just want a good firewall that works well with Asterisk. Thanks, David Wathen Depends on what level of firewall you're looking for. For a full firewall on either a dedicated system or one of your own, I cannot strongly enough recommend Astaro Linux firewall. Better throughput than a pix, worlds easier to operate and configure, and comparable in price. Very SIP/VoIP friendly. Loads of optional modules (we use its mail filter module to filter spam/viruses for several hundred thousand user mailboxes, for instance) to limit the cost to what you need. Also has a built in SIP Proxy, although I've never used it. Excellent platform. Of course, at home, I just use a little Linksys WRT box. It's hardly a corporate-grade firewall, but it's quite SIP-friendly. N. No votes for Vyatta? I have been seriously checking it out. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Billing applications
A2billing (Star2Billing, I think, for commercial support) is a good choice and it's very mature software. Astercc is very fast and has a very good callshop solution. Regards, Juan voip crazy wrote: Hello all, I want to instal a Billing solution in the same asterisk's box. I have browse for ast2bill asterisk billing, astercc, and more, bu ti do not know which will be the best for me. The only things i need, are, - Postpaid and prepaid applications. - True CDR. Better that asterisk one, With suport for transfers - I do not need support for reseller - Billing for Voip, PSTN trunks I need a light app. I'm not searching a heavy app. with a lots of modules and applicacions. I need a ligth application for a soho and its needs. Any one are using a billing application which fits this needs? Any clue will be welcomed. Thanks in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Hard Phone with SMS
Does any one know about a SIP hard phone capable of sending SMS messages (Or a SIP MESSAGE) that could be read from Asterisk dial plan?? Or at least with J2ME support, to run a little program? Regards, Juan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CIDlookup
Use CALLERID(name). http://www.voip-info.org/wiki/view/Asterisk+func+callerid Steve Totaro wrote: On Thu, Jul 9, 2009 at 3:01 AM, Sriramd_r_sri...@hotmail.com wrote: Hi List I've a CID lookup hooked onto an inbound route (i m using trixbox) it runs well but it returns the value as CIDNAMECIDNUMBER ... if i just want to display the CIDNAME [leaving the quotes and CIDNUMBER] .. how can i do it ? do i have to edit some macro in extensions.conf ? rgds Sriram ___ Use Cut() http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cut ___ -- 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] Variable using AMI
Maybe you could use the Asterisk Database. In 1.4 you can do it with DBGet and DBPut: http://www.voip-info.org/wiki/view/Asterisk+cmd+DBget http://www.voip-info.org/wiki/view/Asterisk+cmd+DBput In 1.6 use DB() function. Regards, Juan David Backeberg wrote: On Mon, Jul 6, 2009 at 11:37 AM, Carlos Ruiz Diazcarlos.ruizd...@gmail.com wrote: Hello, if I do a variable assignation using AMI interface, that variable will be visible only for the current AMI instance or will be readable for all AMI instances?. I will login using the same user, concurrently. A program will write a global variable using the same name and if asterisk don't have any scope rules I have to find another way to do what I want. If you want to maintain scope for a variable across multiple calls you should maintain the value of that variable outside of asterisk and keep setting it for each new phonecall. Global variables in asterisk do not do what you are describing. AMI does have something where you can name a particular AMI session, and then communication for that session will care that name. That should not be confused with a system-wide global variable. ___ -- 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] Variable using AMI
Well, I do not understand very well what you are trying to do, but I'll give you some advice: If you want a variable only for the AGI you call, you just have to declare that variable on the AGI. If you would like to make visible that variable as long as the call is active and for each call, even if the name is the same, you have to set a channel variable with the Set(variable=value) command. If you would like to have a variable shared between two or more channels, use the SHARED() funcion(Asterisk 1.6, back ported to 1.4) If you want a variable to be accessed from all the channels, you could use a global variable. http://www.voip-info.org/wiki/view/Asterisk+variables Regards, Juan Carlos Ruiz Diaz wrote: Thank you! I did not know the existence of DB command. The command allows me to store KVPs but I have to use the same variable name every time so every process that starts the AMI instance will override the values making it unusable for what I want to achieve. It was really useful anyways. :) 2009/7/6 Juan E. Rodríguez jerdg...@gmail.com mailto:jerdg...@gmail.com Maybe you could use the Asterisk Database. In 1.4 you can do it with DBGet and DBPut: http://www.voip-info.org/wiki/view/Asterisk+cmd+DBget http://www.voip-info.org/wiki/view/Asterisk+cmd+DBput In 1.6 use DB() function. Regards, Juan David Backeberg wrote: On Mon, Jul 6, 2009 at 11:37 AM, Carlos Ruiz Diazcarlos.ruizd...@gmail.com mailto:carlos.ruizd...@gmail.com wrote: Hello, if I do a variable assignation using AMI interface, that variable will be visible only for the current AMI instance or will be readable for all AMI instances?. I will login using the same user, concurrently. A program will write a global variable using the same name and if asterisk don't have any scope rules I have to find another way to do what I want. If you want to maintain scope for a variable across multiple calls you should maintain the value of that variable outside of asterisk and keep setting it for each new phonecall. Global variables in asterisk do not do what you are describing. AMI does have something where you can name a particular AMI session, and then communication for that session will care that name. That should not be confused with a system-wide global variable. ___ -- 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 ___ -- 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] PHP AGI Not Working and Odd Behavior
Try running your script with /usr/bin/php5 script.php to test it Or changing #!/usr/bin/php5 -q to #!/usr/bin/php -q Leah Newmark wrote: Thanks. I didn't change anything in my dialplan. I am aware of reloading configuration :) My AGIs are copied from a working asterisk install -- the shebang argument is how I've always done it. Either way, I have tried it without the -q as well, and that also didn't succeed. I just tried your test and it worked fine to run it. As I said, I know the server is reading the file I've been editing. I see it on the monitor. It's definitely opening the file to return that message... Any other ideas? On Wed, Jun 24, 2009 at 3:43 PM, Leah Newmarklnewmark at capalon.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote: / Hi, // // I'm running asterisk 1.4.22 on a debian server. // I have php5 installed and it works correctly command line. // When trying to run a php script via AGI, I get messages such as: // GI Tx I // AGI Rx #!/usr/bin/php5 -q // AGI Tx 510 Invalid or unknown command // // The scripts are completely executable and owned by asterisk // -rwxr-xr-x 1 asterisk asterisk // // Googling is not helping much, and php seems installed perfectly. Other // servers running the same AGIs have no such problem. // // I also have noticed odd behavior. When I edit an AGI, the changes aren't // always showing up in the running of the script via asterisk. // For example, I tried editing the bash command to read #!/usr/bin/php -q, // and got the same response on my agi debug. // I know for a fact it's running the script I've edited: // Launched AGI Script /var/lib/asterisk/agi-bin/myscript.php // and it's not in some other directory. / Keep in mind that if you change your dialplan to call a different script you will need to cli dialplan reload Other than that, I'm not sure that it's legal to put an argument into a shbang, as in your -q when launching php. It's also possible you've somehow locked down php or directories way too much. The proper test is to: bash$ sudo -u asterisk /path/to/script.php ___ -- 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 SIP calls stop working having more than 300 calls (more than 600 channels)
I do, I am planning to have little more than 1000. Right now I had managed little more than 700 SIP channels + 100 IAX channels. Do you think this can cause any problem?? --I mean, having this RTP ports range-- Tzafrir Cohen wrote: On Fri, Oct 17, 2008 at 01:24:35AM -0400, Juan Rodríguez wrote: Tzafrir: Following the comments on your post, I started checking (after breaking my head 'googling') the UDP ports in use, and found out that the script that my Asterisk is running was using UDP connection too. This caused that ports from 10,000 to 20,000 could not be used by Asterisk. I change the port range from 10,000 to 40,, and now everything looks OK. Why not change it to 9000- ? Do you actually need more than 1000 sockets at a time? ___ -- 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] 1 second delay when connecting calls
Neal: Try having on sip.conf: srvlookup=no Regards, Juan [EMAIL PROTECTED] wrote: Hello, Thanks for your replies. We checked our sip.conf and we have canreinvite=no already. I agree it could be a firmware issue. I will get another vendors phone hooked up to the pbx before going crazy with support. Thanks, Neal On Sun, Oct 12, 2008 at 6:14 AM, Vieri [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: --- On Sat, 10/11/08, Eric ManxPower Wieling [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Try setting canreinvite=no in each of the device sections on a couple of phones, reload chan_sip.so and see if that fixes things. It has fixed the issue when I've tried it. [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello, We are using asterisk 1.6, sangoma pri card, and Cisco 7960 phones. When we make or receive calls there is a delay before voice is heard. Anyone have any ideas on where to start to debug or has anyone seen this before. We have played with settings on pri, asterisk, and phones with no change. I'm having the same problem but with ATA-connected analog phones. The ATAs are Grandstream GXW4008 with firmware v. 1.0.1.15 http://1.0.1.15. The canreinvite option in sip.conf doesn't change anything for me. Downgrading the GXW4008 solves this issue so this is obviously a firmware bug in my case. I had a vague report once of a user in another installation having this 1-second delay on call connection. That user had a Cisco phone but I don't remember which one. I suggest you check this with Cisco Support if you can. Vieri ___ -- 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 ___ -- 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
