Re: [asterisk-users] Provider UNREACHABLE
On Jul 11, 2006, at 11:10 AM, Rick Smith wrote: teliax had a 2.5 hour outage today. I wouldn't call that short. They notified about the situation when it happened and explained it when it ended, so that makes them look extremely professional compared to other vendors in this department. Marty ___ --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
Re: [asterisk-users] Server redundancy
You said there are 3 asterisk servers in your system. Are you using ARA? Is it a multi-asterisk configuration? Do you mean to tell the configuration of your environment? On 7/12/06, Douglas Garstang [EMAIL PROTECTED] wrote: Assuming a standard usage rate of 10-15%, that's 2,400 concurrent users. If you divide that by 120, that's about 20 Asterisk systems. We're no where near that yet, and we only have three systems up right now. Gotta start somewhere! ___ --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
Re: [asterisk-users] Server redundancy
For redundancy on the PRI side, we plug our PRIs into Redfone Networks' foneBRIDGE boxes. They've worked quite well for us. As others have stated, use short registration periods combined with some HA software to handle your SIP redundancy. You might also look into load-balancing SER proxies. On 7/10/06, Alejandro Acosta [EMAIL PROTECTED] wrote: Hi all, I have read a lot about * server redundancy, however I still don't know how to do it. Can any of you give me any advise? For example, I've read about ranchnetworks appliances but don't know if it will solve my problems. As you may guess, I need to have two servers with the same information (including configuration, cdrs and logs). Of course, If a server fells down I would like the IP Phone to register with the other server. Can I do that?, do I need a third server?, other appliances? Alejandro Acosta, ___ --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/asterisk-users
Re: [asterisk-users] Problem of Quality
On Jul 11, 2006, at 2:27 AM, Olivier Saulnier wrote: Hello, Sometimes, when i call an outside people, he said me that the communication is bad: The voice is low, far, bad poor quality. How can i know where is the problem, which tests can i make? Make sure the string is tight between the two tin cans. But seriously you might try actually giving us some info if you want help? ___ --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
Re: [asterisk-users] NuFone, please send the log file
On Jul 11, 2006, at 5:14 PM, Ronald Wiplinger wrote: That is easy to calculate: 3,000 US$ times your zip code times the phone number you are calling times 2.9cents/5 seconds divided by the Social Security number of the called party ... Or how does NuFone calculate that? But hey, just look at the log file, hmm, didn't we start here? WHERE ARE THE LOG FILES Thanks for all the encouraging funny answers. I go now to 7-eleven to buy some candies, ... Ask if they have any thorazine while you are there... ___ --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
Re: [Asterisk-Users] dtmfmode=inband but SDP also indicates rfc2833
On Jul 11, 2006, at 5:44 PM, Kevin P. Fleming wrote: - Stagg Shelton [EMAIL PROTECTED] wrote: I did ultimately force asterisk to the point where it will not accept or send rfc2833. I did this by modifying chan_sip.c in the function Asterisk should not be sending an SDP with RFC-2833 in it when the dtmfmode=inband in sip.conf. If it is doing that, please capture a 'sip debug' of this happening and opening a bug on bugs.digium.com :-) Glad to hear that. That's what I figured, but he was too busy hacking his code to check... ___ --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
Re: [Asterisk-Users] flash button on asterisk + legacy pbx system
Thank you very much!!!
Giorgio Incantalupo
C F wrote:
Yes you can do that just change the application map to a Goto command
that goes to an exten in the dialplan that does it all for you.
On 7/11/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi C F,
I managed to send the flash code...thanks for your help. Now I'm trying
to send digits after the flash code so that the user can send another
extension. Is it possible to have something like
key = _X.,caller,SendDTMF,X.
inside [applicationmap] in order to send digits to the legacy pbx?
TIA
Giorgio Incantalupo
C F wrote:
Yes I have seen this before and it creates confuseion, but the
solution is that you create 2 application maps, one that works for
inbound calls, and the other that works for outbound calls.
The following is what works for me:
/etc/asterisk/features.conf:
[applicationmap]
inflash = *4,caller,Flash,()
outflash = *3,callee,Flash,()
/etc/asterisk/extensions.conf:
exten = s,1,Set(DYNAMIC_FEATURES=inflash);this is an incoming call on
the FXO port and g2 are the FXS ports
exten = s,2,Dial(Zap/g2,,t)
exten = _1XX,1,Set(DYNAMIC_FEATURES=outflash);this is
outbound
exten = _1XX,2,Dial(Zap/g2/${EXTEN},,T)
With the above they dial *4 on incoming calls, and *3 on outgoing
calls to get this working.
I know it's confusing, but the users get used to it.
On 7/4/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi C F,
ok, I also thought to make the user to press some keys for example
* and
3 so I setup a little test made using an Asterisk box with a
TDM400P (2
FXS + 2 FXO) connected to an analog phone (fxs port) and an analog
line
(fxo port).
I searched on internet and found some interesting stuff so I made my
extensions.conf:
My extension.conf is (in brief):
[zap]
exten = s,1,Set(DYNAMIC_FEATURES=zapflash)
exten = s,2,Dial(Zap/3,15,tw) --- Zap/3 is my analog phone
exten = s,3,HangUp
My zapata (Zap/1 is the line and Zap/3 is the phone):
context = zap
language = it
signalling = fxs_ks
threewaycalling=yes
transfer = yes
channel = 1
language = it
signalling = fxo_ks
callerid = tel1 100
threewaycalling=yes
transfer = yes
channel = 3
and my features.conf:
[applicationmap]
...
zapflash = *3,caller,flash,()
When I call the number xxx, Asterisk answers on zap line
passing the
call to zap/3. I pick up zap/3 phone and then I press *3 but all I
get
is (on asterisk console):
WARNING[3082]: app_flash.c:101 flash_exec: Zap/3-1 is not an FXO
Channel
Why? It seems Asterisk sends Flash command to the phone but it is not
what I want.
Is this the right way to follow? Press *3 (or other code) to send
command to host pbx while the callee is on the phone? Is this what
you
meant? If yes, why Asterisk does not send the flash command to the
line?
Thanks for patience
Giorgio Incantalupo
C F wrote:
Sorry I didn't realize this is how you wanted it to work - that the
user is on a FXS and you want when the user flashes that it flashes
the host pbx.
I disagree with you on this setup the user should be requried to
press
some DTMF and not just flash the phone. The main reason being that
otherwise you will lose 3way and callwaiting features on
asterisk. I'm
assuming your answer to this is that you don't care since you just
want to make the phone an extended extension on the host PBX,
and want
it to be as much an extension of the old PBX as posible. I still
disagree because as much as you are going to try, your users will
still not see this as a direct extension, and sooner or later
you/they
will have to learn how to deal with it anyhow.
On 7/4/06, Giorgio Incantalupo [EMAIL PROTECTED]
wrote:
Hi C F,
I read the comments but the problem remains...after some tests, I
changed some parameters inside zapata.h and recompiled to make
flash
button work so now my asterisk knows when the user presses the
flash
button /during a call./
My problem now is how to transfer the flash signal to the old
PBX,
infact seems like asterisk accept it (even if I cannot use it
inside
extensions.conf for example with a _FLASH,1,...) but then doesn't
re-send it to the line.
TIA
Giorgio Incantalupo
C F wrote:
Use features.conf,
look here at the comments:
http://www.voip-info.org/wiki-Asterisk+cmd+flash
On 7/3/06, Giorgio Incantalupo [EMAIL PROTECTED]
wrote:
Hi C F,
you say Flash asterisk command send a flash signal to old
pbx so
that it
sees that command as coming from an analog phone. But since
Flash
is not
a digit, how can I catch it from within asterisk? How can I
tell
asterisk (es inside extensions.conf) to do something whene
receive it
from a phone?
TIA
Giorgio Incantalupo
C F wrote:
The flash command will do just that. However flash only
works on
FXO
ports and not on SIP FXO ATAs, if you use the later then you
will have
to
Re: [asterisk-users] New Asterisk server crashes daily
On Jul 11, 2006, at 7:39 PM, Al Lougher wrote: I have 969mb total mem with 780mb allocated as swap ?? ___ --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
[asterisk-users] Urgent call forward
I have 2 context and each have its trunk but when I active call forward the off-net international number goes over the 1st trunk only , How can I solve this problem,I want to let each context goes over its specified trunk * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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
[asterisk-users] IAX2 trunking problems
Hello list We are having some strange problems. When we setup trunking between two of our servers, the connection only uses trunking one way. Ex: Data From callingserver to receivingserver uses trunking Data from receivingserver to callingserver does not use trunking. I discovered this problem by looking at a tcpdump in Ethereal, and I can see that the trunked meta packets only goes one way. The other way uses normal Mini packets with raw a-law data. Heres the configurations, with password, username and server info removed. Callingserver: [gsmgw1] secret=*** username=** host=*** type=peer trunk=yes notransfer=yes disallow=all allow=alaw allow=g726 Receivingserver: [**] secret=*** context=default host=** type=user accountcode= trunk=yes notransfer=yes Both servers have ztdummy module installed and loaded. Regards Jon Schøpzinsky -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.10/385 - Release Date: 11-07-2006 ___ --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
RE: [asterisk-users] Polycom ACD, Asterisk, Kernel 2.6 - now SIP does not register
That worked great, thanks for your help. It has now brought me to another problem if you/anybody can help. When a sip device tries to register with asterisk the CLI comes up with: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.1.3.103' - Username/auth name mismatch The 3002 is the correct extension, 192.1.3.109 is the asterisk server and the .3.103 is the phone. The sip.conf is: [3002] type=friend context=demo secret=3002 username=3002 host=dynamic dtmfmode=rfc2833 mailbox=3002 agentlogin=yes agentcbcontext=demo The auth name and password are setup in the Polycom 501 (I have tried it with the eyebeam software and it does the same thing) Thanks, Dean. -Original Message- From: Tzafrir Cohen [mailto:[EMAIL PROTECTED] Sent: 11 July 2006 23:10 To: Dean @ INKnBITs Subject: Re: [asterisk-users] Polycom ACD, Asterisk, Kernel 2.6 On Tue, Jul 11, 2006 at 10:22:38PM +0100, Dean @ INKnBITs wrote: Re: [asterisk-users] Polycom ACD, Asterisk, Kernel 2.6I'm using Debian 3.1 with kernel 2.6.xx (17 i think) and it does a make ok, but when I do a make install it comes back with unexpected end of file ztcfg (think thats the file, will have to check in the morning at the office). In trunk? Take a look at Makefile in the directory zaptel. Look at the insta: target. Just delete the check for ztcfg and copy it manually, or fix the install cript (there seems to be a missing ';' before the 'fi') -- 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
[asterisk-users] dial plan -- help
I have the following context in the dial plan. in extension.conf [default] 1) context1 2) context2 3) context1,1,Macro(a) 4) context2,1,NoOp 5) context2,2,NoOp [macro-a] 6) exten = s,1, NoOp 7) exten = s,2, MacroExit As I expect the route of a call is 3,6,7,4,5. However, when I execute the plan, it shows me the route as 3,6,7,5. When the macro return to context2, the system will search the second priority instead of the first priority. As a result, some action will miss. In the case above, Anyone can suggest me a good way to do that? Any reset priority command that can be used. Or I misuse in the above case. ___ --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
RE: [asterisk-users] MFC/R2 country and carrier specific protocol variants
Hi Dennis,Thank you for the information. We haven't had that problem so far, in the tests that we have been making (we are not yet in production, but will likely be within a short time frame in several countries), so unfortunately I cannot help you in that specific problem. I hope you can solve it quickly, though. Good luck.Best regards, PauloDennis Nacino [EMAIL PROTECTED] skrev: Hi Paulo,I'm from Philippines and here's the protocol variant line I use for our R2 provider (NextelPhilippines) protocolvariant=ph,12,18,1But it never reach production stage pending resolution of the problem I post in this list last May"[Asterisk-Users] Unicall MFC/R2 B3,B4 and clear back". Anyway, I presumed you've been usingUNICALL/R2 channel in production. May, I know how did you deal with that problem. Should Ipresumed that since R2 is so variant, somehow you've been spare.Regards,Dennis__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 listTo 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/asterisk-users
[asterisk-users] asterisk + nite affiliates
Hi Guys I have a client call center that has after hours agents. Once the call center closes they forward calls to the night affiliates. These nite operators are not constant and tend to swop with each other and then let the person in charge know who is on when. I have the mammoth (Mannie) task of getting a gui solution for asterisk. I would like to open a web browser and have a little drop down box where i can select the after hours workers number (be in home or mobile numbers) and then dial the numbers when incoming calls start hitting the system. They are currently using the Avaya Definity for this, but we are phasing it out. I am running on Suse 10.0 and asterisk 1.2.9.1 Is there already a software out there that can do this or should i have already started coding. Any help will be greatly appreciated. Cheers Terry ___ --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
[asterisk-users] Urgent context
Since you make call forward the forwarded number will be dual by default from the context from-internal which is linked to a trunk How can I let it find the context ? automatically $context ? Please help * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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
[asterisk-users] Possible polycom_acd_functions BUG
I have noticed a couple of issues, unless I'm doing something wrong? I pulled with svn the svn.digium.com/svn/asterisk/team/bweschke/polycom_acd_functions/ which got release 37416 This complies fine, in particular the meetme app. If I setup a sip device in the sip.conf with a username and password, I get: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.1.3.103' - Username/auth name mismatch with out the username and password, I get a registration. (This doesn't help as I need the password field for the ACD function to work) I have gone backwards through the releases, 30432 complies fine, except it will not compile the meetme app, but the username and passwords works fine. Does anybody know a release in the middle that works with both features? Thanks, Dean Bath. ___ --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
[asterisk-users] Queue menu
Hi all, I'm trying to setup a menu in a queue, something like All our agents are busy, press 1 to leave a voicemail, press 2 for another department etc. Anyway, the only thing I found is this on the wiki: Menu for the user You can define a menu for the user, while waiting. For this menu, you can only use one-digit extensions. Define the context for the menu in the configuration for the queue to enable this option. But I don't know what this meanse or atleast how to implement it. Could someone give me a hint in the right direction ? Regards, Attilla ___ --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
[asterisk-users] Midnight Naughties
Hi all, any one know about :Midnight Naughties. The Asterisk box seems to log all agents off Queues at exactly midnight each night, and nothing in cron jobs. Thanks James ___ --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
[asterisk-users] hardware question (X100P X101P clones)
Hi guys, I would like to know if the Ambient and Intel FA82537EP are X100P/X101P compatibles. Thanks Cheers Alex ___ --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
Re: [asterisk-users] Queue menu
Hi, In the queue.conf when you define the queue: [myqueue] context = thecontextforusers ... and in your extensions.conf [thecontextforusers] exten = 1,1,NoOp(1 WAS PRESSED) exten = 2,1,NoOp(2 WAS PRESSED) If you need more help, just ask ! ;) Attilla De Groot a écrit : Hi all, I'm trying to setup a menu in a queue, something like All our agents are busy, press 1 to leave a voicemail, press 2 for another department etc. Anyway, the only thing I found is this on the wiki: Menu for the user You can define a menu for the user, while waiting. For this menu, you can only use one-digit extensions. Define the context for the menu in the configuration for the queue to enable this option. But I don't know what this meanse or atleast how to implement it. Could someone give me a hint in the right direction ? Regards, Attilla ___ --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/asterisk-users
[asterisk-users] SPA-3000 XML Config File
Hi Asterisk Users, Sorry if this is off Topic for this list. But does anyone have a full XML config file for the SPA-3000, the PAP2 and the SPA-941. Or alternatively a way to convert the field names on the web pages to the corresponding XML filed names. Thanks /S ___ --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
Re: [asterisk-users] Queue menu
On Jul 12, 2006, at 1:01 PM, Tristan wrote: Hi, In the queue.conf when you define the queue: [myqueue] context = thecontextforusers ... and in your extensions.conf [thecontextforusers] exten = 1,1,NoOp(1 WAS PRESSED) exten = 2,1,NoOp(2 WAS PRESSED) If you need more help, just ask ! ;) Hi Tristan, Perfectly clear now and it works, thank you. regards, Attilla ___ --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
[asterisk-users] Urgent context
Since you make call forward to an extension l by default it will attach your DIAL(local/[EMAIL PROTECTED]) from the context from-internal which is linked to a trunk , The script is at /var/lib/asterisk/agi-bin/dialparties.agi I $dialstring = 'Local/'.$extnum.'@from-internal'; How can I let it find the context ? automatically $context ? Instead of '@from-internal' Please help regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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
Re: [asterisk-users] SPA-3000 XML Config File
Sorry if this is off Topic for this list. But does anyone have a full XML config file for the SPA-3000, the PAP2 and the SPA-941. Or alternatively a way to convert the field names on the web pages to the corresponding XML filed names. The spa provisioning guide outlines the xml-like syntax. The guide along with a configuration compiler is available from sipura/linksys, but one has to be approved by them to gain access to those tools. The compiler will also generate a complete list of all parameter names and default values, which can be exported to a text file. A short sample looks like this: # *** Configuration Profile Provision_Enable Yes ; Resync_On_Reset Yes ; Resync_Random_Delay 2 ; Resync_Periodic 3600 ; Resync_Error_Retry_Delay 3600 ; Forced_Resync_Delay 14400 ; Resync_From_SIP Yes ; Resync_After_Upgrade_Attempt Yes ; Resync_Trigger_1 ; Resync_Trigger_2 ; Resync_Fails_On_FNF Yes ; Profile_Rule /spa$PSN.cfg ; Profile_Rule_B ; Profile_Rule_C ; The field names provided from the above list can then be used as xml-style parameters. Not sure why sipura needs to lock down access to the guide (and compiler), but they have since day one. The guide that I have (for the spa3k) is an older version that includes 56 pages in the pdf file. I don't know if a similar guide exists for the 941/942. Contact [EMAIL PROTECTED] to get approval and access. ___ --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
[asterisk-users] Re: Re: TE420P/TE415P?
Question rephrase: If I have a sip to voicemail call that needs G.729 transcoding, can it use the Digium hardware transcoder or would I still need a software transcoding license for this? -- -- Steven http://www.glimasoutheast.org C F [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Has the TC400B been released yet? On 7/11/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: - Steven [EMAIL PROTECTED] wrote: I assume that it would be 30 licenses, so you could fully use the card as E1. Is this correct? Can asterisk use these licenses for other calls as well? (sip G.729 to voicemail) Your questions don't make much sense. The TC400B includes all the licenses needed for every call it can handle, regardless of how many that ends up being. There would be no advantage for the licenses to be used for 'other' calls, since if there is a license available the call should be sent through the hardware transcoder anyway. -- Kevin P. Fleming Senior Software Engineer 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 ___ --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/asterisk-users
[asterisk-users] IVR with LDAP query for phone number and mobile number??
We still have a lot of users on a legacy PBX, so the Directory app is not sufficient. We also have users with mobile phones. Has anyone made an LDAP lookup that will pull this info from MS Active Directory? My thinking is to add this function to my main IVR. As long as my AD is accurate, it should contain all my info. Replace the current directory app in my main IVR with this function. When someone presses 9 in my IVR, do a lookup to AD with all of the possible number combinations. ex. 222 is aaa aab aac aba abb abc, etc. I am not sure how I would do the desk phone vs. mobile phone number options. Submit the accepted number to the dialplan so that 4 local digit extensions dial local, 4 digit legacy extensions dial out that PRI and mobiles dial out to PSTN or GSM gateway. Anyway, I haven't fully thought it through, but I figured I would ask if anyone else had done this yet. -- -- Steven http://www.glimasoutheast.org ___ --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
Re: Re: [Asterisk-Users] app_conference DTMFs?
On 7/12/06, Henry J. Cobb [EMAIL PROTECTED] wrote: When I've tried it, app_conference always crashed within the hour. that's strange. we've use app_conference for months and months on end without incident.are you building app_conference from the main svn trunk? or are you using matt's VD_app_conference that he mentioned a couple posts ago? j- ___ --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
Re: [asterisk-users] Yet another problem with incoming SIP calls and 407
Eric \ManxPower\ Wieling [EMAIL PROTECTED] writes: Wolfgang Zweimueller wrote: Hi all, when I receive incoming SIP calls on my Asterisk (1.2.9.1) where the caller has a username in it's From-Address which also exists in my sip.conf then my system answers with 407 Proxy Authentication Required. If it's nonexistent username then callin works fine! It seems that this is a problem in the SIP implementation of Asterisk and found a few hints on how to resolve this (allowguest=yes, insecure=invite,port etc.). But none of them does help! Can anyone suggest what I else could try? in sip.conf [general] context=INVALID Then put the correct context= line for each sip user/friend/peer. Unauthenticated calls use the options in [general] That's already there! Any other ideas? cu, Wolfgang ___ --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
Re: Re: [Asterisk-Users] app_conference DTMFs?
[EMAIL PROTECTED] wrote: On 7/12/06, Henry J. Cobb [EMAIL PROTECTED] wrote: When I've tried it, app_conference always crashed within the hour. that's strange. we've use app_conference for months and months on end without incident. are you building app_conference from the main svn trunk? or are you using matt's VD_app_conference that he mentioned a couple posts ago? I tried several different combinations of app_conference and Asterisk versions and then I had to get back to actually providing phone service that didn't crash. -HJC ___ --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
Re: Re: [Asterisk-Users] app_conference DTMFs?
It really depends on the application. app_conference does wonderfully for long conferences without a lot of entry/exit and no playing of audio files. The issues with the double-free crashes that we've had all seem to be caused by playing of audio files(like the entry/exit sounds or the DTMF broadcast). But these functions use the app_conference code that was already existing to play audio files from manager API commands so the issue was there it just wasn't as tested because not many people use the manager command a lot to play audio in conferences. The other issue we had with app_conference was using it in high-volume VICIDIAL outbound production(thousands of entry/exit actions per hour) where it would always fail after 1-8 hours. In this case there wasn't a crash, but strangely app_conference just seemed to stop working like the engine died. Everything else in Asterisk kept working but you couldn't do anything in app_conference without stopping and starting Asterisk again. MATT--- On 7/12/06, jeff oconnell [EMAIL PROTECTED] wrote: On 7/12/06, Henry J. Cobb [EMAIL PROTECTED] wrote: When I've tried it, app_conference always crashed within the hour. that's strange. we've use app_conference for months and months on end without incident. are you building app_conference from the main svn trunk? or are you using matt's VD_app_conference that he mentioned a couple posts ago? j- ___ --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/asterisk-users
Re: [asterisk-users] OT: 3Com 3C10222 POE 24 Port Ethernet
Erick Perez wrote: There is an old, very old document that I found somewhere that this PoE switch was designed for NBX phones at that time. Does anybody in this list is using this switch with non-3com NBX PoE phones? just check the voltage specs. I think you will fry anything other than an old 3com phone. Now I believe they use the standard PoE in their new switches. ___ --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
[asterisk-users] Problem incoming calls from sipphone/giztmo
Hello When I try to use giztmo whit this configuration i'm unable to receive calls register = 1747601:[EMAIL PROTECTED] externhost= host.homeip.net port=5060 defaultexpirey=3600 localnet= 192.168.0.0/255.255.255.0 But whit this configuration y can receive calls register = 1747601:[EMAIL PROTECTED] externhost=host .homeip.net port=5060 defaultexpirey=3600 But externals users are unable to connect with my asterisk server ___ --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
Re: [asterisk-users] NuFone, please send the log file
Sounds like class action lawsuit time. Michael Workman wrote: So Nufone Screwed ya I feel Sorry... W Take your Lumps... Cut Your Losses and Get on with Life Your not the only one Nufone Screwed They Screwed me Out of $3,000.00 NEXT TIME BEFORE YOU GET SCREWED BUY SOME KY JELLY AND ENJOY IT. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Tuesday, July 11, 2006 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] NuFone, please send the log file Andrew D Kirch wrote: Ronald Wiplinger wrote: Dear NuFone, Without misunderstanding I ask you again, please send the log file and pay back my money! Not following this request results in the assumption that NuFone is cheating and I will post this info every hour on more Internet places. This should help that other people will not trap into a cheating company. bye Ronald ___ --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 I'm going to note two more issues I've just found with this post. 1. this is a specifically NON-Commercial list (your post is commercial) Yes, I did not write too much, but one part of the issue is, that NuFone does not answer to technical questions either, but asks for set-up help in IRC. So to see, it is a hint for technical people to take care if they suddenly get an offer for consulting, just when you ask a technical question. 2. you have threatened to post it to further such lists and forums where it is not desired (your post is being made in bulk) I therefore must determine you have posted UCE/UBE and you are a spammer. I strongly disagree with that! places are not only lists! Maybe you are too new on the net to figure out, that there are still other places. Have you tried to Google for Nufone? Than you might find other places too. Again, I just want to have the log files. I do not get answer and that is a fact. If you have good contacts to Jeremy, maybe you can convince him to send the log file. It is that simple. I have set-up a filter for NuFone, and when I have time and catch a message with that trigger word, I will post my thoughts. Thanks for pointing out not to send too many messages. However, to answer to another ones message, . have a nice day! bye Ronald ___ --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 --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0628-2, 2006/07/11 Tested on: 2006/7/12 ¤W¤È 07:29:36 avast! - copyright (c) 1988-2006 ALWIL Software. http://www.avast.com -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ --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/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/asterisk-users
Re: Re: [Asterisk-Users] app_conference DTMFs?
interesting. i didn't realize the problem seems to specifically be the sound playback via the manager interface.a couple weeks ago, my asterisk on my dev box crashed, i did some preliminary investigation, but since we hadn't had any problems in production or qa, i chalked it up to me messing up my dev install somehow. anyway, here's the backtrace of the core:(gdb) bt#0 0x00df71bb in ?? () from /lib/libgcc_s.so.1#1 0x080625ad in ast_deactivate_generator (chan=0x815fb40) at channel.c:1382#2 0x0806d77a in ast_openstream_full (chan=0x815fb40, filename=0x815dcec tones-exit, preflang=0x0, asis=0) at file.c:494#3 0x0806d835 in ast_openstream (chan=0x71bfb8a6, filename=0x71bfb8a6 Address0x71bfb8a6 out of bounds, preflang=0x71bfb8a6 Address 0x71bfb8a6 out of bounds) at file.c:467#4 0xb7e8899b in conference_play_sound (fd=12, argc=14643636, argv=0xdfdc5c)at cli.c:225#5 0x0809786b in ast_cli_command (fd=12, s=0x71bfb8a6 Address 0x71bfb8a6 outof bounds) at cli.c:1364#6 0x080b2f8a in action_command (s=0x815c448, m=0xb7b97420) at manager.c:927#7 0x080b7b81 in process_message (s=0x815c448, m=0xb7b97420) at manager.c:1305#8 0x080b83cf in session_do (data="" at manager.c:1401#9 0x00b44341 in start_thread () from /lib/tls/libpthread.so.0#10 0x009096fe in clone () from /lib/tls/libc.so.6the interesting lines to me are #4 and #5:#4 0xb7e8899b in conference_play_sound (fd=12, argc=14643636, argv=0xdfdc5c) at cli.c:225 #5 0x0809786b in ast_cli_command (fd=12, s=0x71bfb8a6 Address 0x71bfb8a6 out of bounds) at cli.c:1364 line 4 becauset the argc passed into conference_play_sound() is so large, and line 5 because there seems to be an out--of-bounds problem in the asterisk code ( i.e. before the app_conference code is called ). based on what you said in your last post, i'm going to look at this more.if you have any thoughts on my backtrace/analysis, let me know.j-On 7/12/06, Matt Florell [EMAIL PROTECTED] wrote: The issues with the double-free crashes that we've had all seem to becaused by playing of audio files(like the entry/exit sounds or theDTMF broadcast). But these functions use the app_conference code thatwas already existing to play audio files from manager API commands so the issue was there it just wasn't as tested because not many peopleuse the manager command a lot to play audio in conferences. ___ --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
[asterisk-users] Automatic Hangup problem on IAX2 communication to Asterisk
Hi all, I'm having a problem with receiving calls from a VOIP provider who is providing inbound PSTN termination using IAX2 to my [EMAIL PROTECTED] 2.6 box. The box is a mini-ITX based P5000 system running off a 2.5in drive with a digium TD400P (3 FXO). But this problem does not relate to the card. Basically the provider gets a call in but when he talks to my server the two don't establish a call. At the same time as this happens I am able to access my system using my Digium IAXY. I turned the IAX2 debugging on at the console and this is kind of what I see. The provider gets a call and sends a frame to open the call. I then take 13ms to send a Challenge request back to him. By this time he seems to have sent a retry at 10ms. At this point he then receives the challenge but sends back an INVAL which causes a hangup. This process repeats until the PSTN is hung up. Am I right in thinking there is a timeout here and this is causing the problem? I enclose the debug log. Any help appreciated. I've replace the actual phone numbers and ips for obvious reasons Any Help appreciated, as I seem to have gone off the end of my service providers knowledgebase. Cheers Rajiv -My IAXY device reacknowledging my Asterisk box-(working)--- Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 2ms SCall: 14336 DCall: 0 [IAXY_IP] USERNAME: IAXY EXTENSION REFRESH : 60 DEVICE TYPE : iaxy2 SERVICE IDENT : 0003640011a3 PROVISIONG VER : 0 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 4ms SCall: 3 DCall: 14336 [IAXY_IP] AUTHMETHODS : 3 CHALLENGE : 680869675 USERNAME: IAXY EXTENSION Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 5ms SCall: 14336 DCall: 3 [IAXY_IP] USERNAME: IAXY EXTENSION MD5 RESULT : a79821d4dd5c0adce83b88d3e5e6ed2a DEVICE TYPE : iaxy2 SERVICE IDENT : 0003640011a3 PROVISIONG VER : 0 Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00050ms SCall: 3 DCall: 14336 [IAXY_IP] USERNAME: IAXY EXTENSION DATE TIME : 2006-07-11 16:34:22 REFRESH : 60 APPARENT ADDRES : IAXY_IP MESSAGE COUNT : 512 CALLING NUMBER : IAXY EXTENSION CALLING NAME: device Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00050ms SCall: 14336 DCall: 3 [IAXY_IP] --Dialed 0845... from my mobile(broken) Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00010ms SCall: 00011 DCall: 0 [Service Provider IP] VERSION : 2 CALLED NUMBER : PSTN NUMBER INBOUND CODEC_PREFS : () CALLING NUMBER : CALLER ID OF ORIGINATING CALL CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 LANGUAGE: en USERNAME: Service Provider USERNAME FORMAT : 4 CAPABILITY : 65407 ADSICPE : 2 DATE TIME : 2006-07-11 16:34:24 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00013ms SCall: 4 DCall: 00011 [Service Provider IP] AUTHMETHODS : 3 CHALLENGE : 266133398 USERNAME: Service Provider USERNAME Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 00011 DCall: 4 [Service Provider IP] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00010ms SCall: 00011 DCall: 0 [Service Provider IP] VERSION : 2 CALLED NUMBER : PSTN NUMBER INBOUND CODEC_PREFS : () CALLING NUMBER : CALLER ID OF ORIGINATING CALL CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 LANGUAGE: en USERNAME: Service Provider USERNAME FORMAT : 4 CAPABILITY : 65407 ADSICPE : 2 DATE TIME : 2006-07-11 16:34:24 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00011ms SCall: 5 DCall: 00011 [Service Provider IP] AUTHMETHODS : 3 CHALLENGE : 198315063 USERNAME: Service Provider USERNAME Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 00011 DCall: 5 [Service Provider IP] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP Timestamp: 02920ms SCall: 00011 DCall: 0 [Service Provider IP] CAUSE CODE : 0 Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 0 DCall: 00011 [Service Provider IP] Rx-Frame Retry[Yes] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP Timestamp:
[asterisk-users] context
Since I make call forward to an extension l by default it will attach your DIAL(local/[EMAIL PROTECTED]) from the context from-internal which is linked to a trunk , The script is located at /var/lib/asterisk/agi-bin/dialparties.agi I $dialstring = 'Local/'.$extnum.'@from-internal'; How can I let it find the context ? automatically $context ? Instead of '@from-internal' Please help regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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
Re: [asterisk-users] context
That's the fourth time you've asked the same question in the space of a few hours - please have a little more patience and wait for someone to answer. On 12/07/06, Khaled Chehab [EMAIL PROTECTED] wrote: Since I make call forward to an extension l by default it will attach your DIAL(local/[EMAIL PROTECTED]) from the context from-internal which is linked to a trunk , The script is located at /var/lib/asterisk/agi-bin/dialparties.agi I $dialstring = 'Local/'.$extnum.'@from-internal'; How can I let it find the context ? automatically $context ? Instead of '@from-internal' Please help regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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 -- Peter Bowyer Email: [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
Re: [asterisk-users] AGI tutorials
Thanx for the tips guys. I need one more favour. can anybody tell me where to find help for writing AGI scripts in C language. I have read the pdf book called Asterisk TFOT, but it explains AGI scripting in languages other than C. I feel comfortable using C language, so i didnt understand the concepts fully.Anybody who knows a good source of AGI scripting in C, plz share On 7/11/06, Kai Ober [EMAIL PROTECTED] wrote: Rizwan Hisham schrieb: Anybody who knows a good source of AGI tutorials on the net? plz share try one of the mirrors and then the pages on AGI,http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 have PhunKai___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- RegardsRizwan HishamSoftware Engineer ___ --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
Re: [asterisk-users] Issues with making Transfers
Dan Brummer wrote: This has worked. I downgraded from 1.2.9.1 to 1.2.7.1 and I'm not having the warm transfer issue anymore. Does anyone know if this is a known issue and is going to be fixed in upcoming release? Should I possibly put in a bug request? -Dan I'm pretty sure I saw this in the bugtracker and that it had been fixed in trunk. Someone on here is bound to know more. The weird thing is, I had problems with call transfers at the same time as upgrading to 1.2.9.1, and it turned out that changing the firmware on the handsets fixed it. Mind you, there's bound to be a new release fairly soon, (there's been a lot of bug fixes lately). ___ --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
Re: [asterisk-users] Re: Re: TE420P/TE415P?
Can you explain why this would be different? On 7/12/06, Steven [EMAIL PROTECTED] wrote: Question rephrase: If I have a sip to voicemail call that needs G.729 transcoding, can it use the Digium hardware transcoder or would I still need a software transcoding license for this? -- -- Steven http://www.glimasoutheast.org C F [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Has the TC400B been released yet? On 7/11/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: - Steven [EMAIL PROTECTED] wrote: I assume that it would be 30 licenses, so you could fully use the card as E1. Is this correct? Can asterisk use these licenses for other calls as well? (sip G.729 to voicemail) Your questions don't make much sense. The TC400B includes all the licenses needed for every call it can handle, regardless of how many that ends up being. There would be no advantage for the licenses to be used for 'other' calls, since if there is a license available the call should be sent through the hardware transcoder anyway. -- Kevin P. Fleming Senior Software Engineer 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 ___ --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/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/asterisk-users
[asterisk-users] 1000s of extensions in one context?
Hi, is several 1000s of extensions in a context a problem? Roger. ___ --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
Re: [asterisk-users] Provider UNREACHABLE
Thanks All First off I never mentioned Teliax (but yes correctly ASSUMED they are my provider) and this is not a Teliax issue per se My issue is more the fact that I have Qualify = yes in sip.conf but repeatedly get REACHABLE and UNREACHABLE as can be seen below. even when I set Qualify = 3600 I still get this My question is more (a) how do I stop this ? (b) What is happening ? Thanks all Barry snip... Jul 11 13:10:38 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 57 Jul 11 13:10:48 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (117ms / 2000ms) Jul 11 13:18:01 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 13:24:17 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 2432 Jul 11 13:24:40 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (63ms / 2000ms) Jul 11 14:08:47 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 153 Jul 11 14:08:57 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (66ms / 2000ms) Jul 11 18:12:21 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 52 Jul 11 18:12:31 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (62ms / 2000ms) Jul 11 18:19:44 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:03:02 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:08:07 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 89 Jul 11 19:08:17 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:17:21 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 50 Jul 11 19:17:31 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 23:01:50 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 50 Jul 11 23:02:01 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) snip. ___ --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
[asterisk-users] Problem with making outgoing calls
Hi, We could makecalls to USA using Teliax service upto 11, July 2006 with Asterisk. But, since 11, July 2006 evening, we are unable to make calls sometimes and could not connect to Teliax server sometimes. I have realized that Teliax server was down for few hours. Currently our Asterisk server is connecting with Teliax. But, When I am trying to make call to USA, Its giving me one ring and being disconnected. I could not understand what could be the problem? Isthere any problem with my connection to Teliax server? With the configuration mentioned below, we could successfully make calls to USA. But, from this morning we are unable to make calls to USA. IAX.CONF file contents: disallow=all allow=ulaw SIP.CONF file contents: [101] type=friend username=101 secret=abcd callerid="Ani" host=dynamic context=tutorial [general]register =
ab.cd:[EMAIL PROTECTED] [authentication]auth = ab.cd:[EMAIL PROTECTED][teliax]context=default type=friendusername=ab.cduser=ab.cdhost=voip-co1.teliax.comsecret=xxinsecure=verycanreinvite=nodisallow=allallow=ulawallow=alawallow=gsm EXTENSIONS.CONF file contents: exten = 101,1,Dial(SIP/101,15) exten = 101,2,Voicemail(u101) exten = 101,3,Voicemail(b101) exten = 101,4,Hangup exten = _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr)exten = 3031234567,1,Answer()exten = 3031234567,2,DIAL(SIP/user,20) VOICEMAIL.CONF file contents: 101 = ani, Ani, [EMAIL PROTECTED], [EMAIL PROTECTED] Please let me know the problem ASAP. Looking forward to your response. Thank you.Regards,Chandra.
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Re: [asterisk-users] Problem with making outgoing calls
On 12/07/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi, We could make calls to USA using Teliax service upto 11, July 2006 with Asterisk. But, since 11, July 2006 evening, we are unable to make calls sometimes and could not connect to Teliax server sometimes. I have realized that Teliax server was down for few hours. Currently our Asterisk server is connecting with Teliax. But, When I am trying to make call to USA, Its giving me one ring and being disconnected. I could not understand what could be the problem? Is there any problem with my connection to Teliax server? What did Teliax support say? I presume they were your first port of call, since they're the people prividing you with service -- Peter Bowyer Email: [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
Re: [asterisk-users] 1000s of extensions in one context?
several thousand extensions or several extensions called 1000 ? - Original Message - From: Roger Schreiter [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, July 12, 2006 9:15 AM Subject: [asterisk-users] 1000s of extensions in one context? Hi, is several 1000s of extensions in a context a problem? Roger. ___ --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/asterisk-users
Re: [asterisk-users] Provider UNREACHABLE
do what we all do. get backup routes - Original Message - From: Barry Fawthrop [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 12, 2006 9:15 AM Subject: Re: [asterisk-users] Provider UNREACHABLE Thanks All First off I never mentioned Teliax (but yes correctly ASSUMED they are my provider) and this is not a Teliax issue per se My issue is more the fact that I have Qualify = yes in sip.conf but repeatedly get REACHABLE and UNREACHABLE as can be seen below. even when I set Qualify = 3600 I still get this My question is more (a) how do I stop this ? (b) What is happening ? Thanks all Barry snip... Jul 11 13:10:38 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 57 Jul 11 13:10:48 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (117ms / 2000ms) Jul 11 13:18:01 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 13:24:17 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 2432 Jul 11 13:24:40 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (63ms / 2000ms) Jul 11 14:08:47 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 153 Jul 11 14:08:57 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (66ms / 2000ms) Jul 11 18:12:21 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 52 Jul 11 18:12:31 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (62ms / 2000ms) Jul 11 18:19:44 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:03:02 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:08:07 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 89 Jul 11 19:08:17 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:17:21 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 50 Jul 11 19:17:31 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 23:01:50 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 50 Jul 11 23:02:01 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) snip. ___ --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/asterisk-users
[asterisk-users] Asterisk version: 1.2.9.1 or older?
Hello, I need to install Asterisk on a test machine that will soon become a production environment. Do you think that 1.2.9.1 is reliable? I read some posts that say it isn't as good as the previous versions. Should I install 1.2.8 or 1.2.7.1? Please give me an advice! Thanks in advance, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --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
[asterisk-users] waitexten only provides one digit in chan_zap
Hi, I want to implement a lookup for valid extensions using agi. Thus I want chan_zap to accept some digits, then check via agi if the number is complete, run waitexten if necessary and check again ... Unfortunately waitexten only accepts one digit, regardless how may key strokes I did on my phone set. Even if I jump back in the dialplan, for asterisk passes again at the waitexten command, no more digits are accepted. Is waitexten no the right command to execute with chan_zap, an E1 line and overlap dialling? Is there something similar to misdn's waitfordigits, which I could use together with chan_zap? Roger. ___ --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
Re: [asterisk-users] Provider UNREACHABLE
(b) What is happening ? If i unplug network cable from my ipphone asterisk will say UNREACHABLE after few seconds. Sometimes it occurs for my sip provider. I simply lose connection with my phone/provider or connection is so poor that transmission of voice will be very choppy, so beeter not to start a call at all. Mayby someone download big files in Your network and take all bandwitch ? for me it's a network issue. -Filip Użytkownik Barry Fawthrop napisał: Thanks All First off I never mentioned Teliax (but yes correctly ASSUMED they are my provider) and this is not a Teliax issue per se My issue is more the fact that I have Qualify = yes in sip.conf but repeatedly get REACHABLE and UNREACHABLE as can be seen below. even when I set Qualify = 3600 I still get this My question is more (a) how do I stop this ? (b) What is happening ? Thanks all Barry snip... Jul 11 13:10:38 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 57 Jul 11 13:10:48 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (117ms / 2000ms) Jul 11 13:18:01 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 13:24:17 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 2432 Jul 11 13:24:40 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (63ms / 2000ms) Jul 11 14:08:47 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 153 Jul 11 14:08:57 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (66ms / 2000ms) Jul 11 18:12:21 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 52 Jul 11 18:12:31 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (62ms / 2000ms) Jul 11 18:19:44 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:03:02 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:08:07 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 89 Jul 11 19:08:17 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:17:21 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 50 Jul 11 19:17:31 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 23:01:50 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 50 Jul 11 23:02:01 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) snip. ___ --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
Re: [asterisk-users] Problem with making outgoing calls
Crazy Boy wrote: Hi, We could make calls to USA using Teliax service upto 11, July 2006 with Asterisk. But, since 11, July 2006 evening, we are unable to make calls sometimes and could not connect to Teliax server sometimes. I have realized that Teliax server was down for few hours. snip Please let me know the problem ASAP. Looking forward to your response. Here's an idea! Contact Teliax support. I have had no problem making or receiving calls. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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
Re: [asterisk-users] Problem with making outgoing calls
Hi,Thank you for response. I sent an email to Teliax people also. I may get reply from Teliax within few hours. Please tell me the solution. Thank you.Regards,Chandra.Peter Bowyer [EMAIL PROTECTED] wrote: On 12/07/06, Crazy Boy <[EMAIL PROTECTED]> wrote: Hi, We could make calls to USA using Teliax service upto 11, July 2006 with Asterisk. But, since 11, July 2006 evening, we are unable to make calls sometimes and could not connect to Teliax server sometimes. I have realized that Teliax server was down for few hours. Currently our Asterisk server is connecting with Teliax. But, When I am trying to make call to USA, Its giving me one ring and being disconnected. I could not understand what could be the problem? Is there any problem with my connection to Teliax server?What did Teliax support say? I presume they were your first port ofcall, since they're the people prividing you with service-- Peter BowyerEmail: [EMAIL PROTECTED]___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail Beta.___ --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
RE: [asterisk-users] Provider UNREACHABLE
It's the internet...maybe for you the path to Teliax is kinda crappy? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry Fawthrop Sent: Wednesday, July 12, 2006 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Provider UNREACHABLE Thanks All First off I never mentioned Teliax (but yes correctly ASSUMED they are my provider) and this is not a Teliax issue per se My issue is more the fact that I have Qualify = yes in sip.conf but repeatedly get REACHABLE and UNREACHABLE as can be seen below. even when I set Qualify = 3600 I still get this My question is more (a) how do I stop this ? (b) What is happening ? Thanks all Barry snip... Jul 11 13:10:38 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 57 Jul 11 13:10:48 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (117ms / 2000ms) Jul 11 13:18:01 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 13:24:17 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 2432 Jul 11 13:24:40 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (63ms / 2000ms) Jul 11 14:08:47 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 153 Jul 11 14:08:57 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (66ms / 2000ms) Jul 11 18:12:21 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 52 Jul 11 18:12:31 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (62ms / 2000ms) Jul 11 18:19:44 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:03:02 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:08:07 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 89 Jul 11 19:08:17 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:17:21 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 50 Jul 11 19:17:31 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 23:01:50 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 50 Jul 11 23:02:01 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) snip. ___ --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/asterisk-users
Re: [asterisk-users] Asterisk version: 1.2.9.1 or older?
Hello, I need to install Asterisk on a test machine that will soon become a production environment. Do you think that 1.2.9.1 is reliable? I read some posts that say it isn't as good as the previous versions. Should I install 1.2.8 or 1.2.7.1? Please give me an advice! Thanks in advance, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --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 Hi I had stability issues with queues on 1.2.9.1. 1.2.7.1 also has queue issues, but it is a LOT more stable. Garth ___ --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
RE: [asterisk-users] Asterisk version: 1.2.9.1 or older?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrea Spadaccini Sent: Wednesday, July 12, 2006 9:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk version: 1.2.9.1 or older? Hello, I need to install Asterisk on a test machine that will soon become a production environment. Do you think that 1.2.9.1 is reliable? I read some posts that say it isn't as good as the previous versions. Should I install 1.2.8 or 1.2.7.1? I would suggest 1.2.9.1 as it is a security update release (ie something that can compromise your PBX is fixed); however the correct answer to this is that you need to test and determine what suits your needs. Andrew ___ --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
[asterisk-users] Echo on PRI
Hi All I have an Asterisk server with two PRI's into a Samsung DCS500 which in turn has 4 PRI's into the Telco. I have echo on some calls only over 500km away. I have set: echocancel=yes echotraining=800 echocancelwhenbridged=yes rxgain=1 txgain=-1 Volumes on the ZAP channels are set correctly. Echo problems in the past have been straight forward to remove with the correct echotraning and volumes set. I am quite certain it is due to calls going through the Samsung. Any ideas? Kind Regards Garth ___ --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
Re: [asterisk-users] asterisk + nite affiliates
I just finished building a gtk gui that's geared toward call centers. Check it out. You're of course welcome to help me code more for it if you like it :) amsuite.sourceforge.net j On Wed, 2006-07-12 at 11:39 +0200, Terry Wade wrote: Hi Guys I have a client call center that has after hours agents. Once the call center closes they forward calls to the night affiliates. These nite operators are not constant and tend to swop with each other and then let the person in charge know who is on when. I have the mammoth (Mannie) task of getting a gui solution for asterisk. I would like to open a web browser and have a little drop down box where i can select the after hours workers number (be in home or mobile numbers) and then dial the numbers when incoming calls start hitting the system. They are currently using the Avaya Definity for this, but we are phasing it out. I am running on Suse 10.0 and asterisk 1.2.9.1 Is there already a software out there that can do this or should i have already started coding. Any help will be greatly appreciated. Cheers Terry ___ --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 -- Credulous at best your desire to believe in angels in the hearts of men Pull your head on out, your head please and give a listen Shouldn't have to say it all again The universe is hostile, so impersonal Devour to survive... so it is, so it's always been We all feed on tragedy --- Tool, Vicarious ___ --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
Re: [asterisk-users] Asterisk version: 1.2.9.1 or older?
I personally have had some issues with 1.2.9.1 in production and had to revert to an older version. We are using 1.2.6 which has proven to be pretty stable. Others might have different experiences. j On Wed, 2006-07-12 at 15:31 +0200, Andrea Spadaccini wrote: Hello, I need to install Asterisk on a test machine that will soon become a production environment. Do you think that 1.2.9.1 is reliable? I read some posts that say it isn't as good as the previous versions. Should I install 1.2.8 or 1.2.7.1? Please give me an advice! Thanks in advance, ___ --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
[asterisk-users] Option D in dial doesnt seem to be working
I have 2 SIP phones ext 401 and 402 and am using 1.2.9.1 (this seemed to work in previous versions). In my dialplan I have exten = 55,1,Dial(SIP/402,,tTD(222)) When 401 comes offhook and dials 55 and I answer on 402 I do not hear the DTFM of 222. Am I doing something wrong? when I change it to exten = 55,1,Dial(SIP/402,,tTD(222:333)) I do hear the DTMF tones on my phone (401) but still not on 402? Jerry ___ --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
[asterisk-users] Lets All Get Smart...
Bell Canada is going to give Canada Wide Calling free in Next few months and Skype is Giving out North American Calls Free until it fixes the security hole in software/system and VoIP Discount is giving out free North American Calling. Anyone who is trying to Sells / Provides VoIP is and already being hurt by this. I figure if we Team up it would not only make our Cost Cheaper but it would help all of us though the hump. What I suggest is we group up and link via DunDi and share a part of PRI's I am willing to make 15 channels available to free calling over PRI's. I suggest we set a Minimum DunDi link requirement - G729 / G723 Codec - 4 Channels Minimum - DSL/Cable Internet Right now we already have 35 Channels of Free Calling linked in and I think if we worked together it would offset our bottom line. if you want to link Send me an Email. ___ --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
Re: [asterisk-users] Asterisk version: 1.2.9.1 or older?
Andrea Spadaccini wrote: Hello, I need to install Asterisk on a test machine that will soon become a production environment. Do you think that 1.2.9.1 is reliable? I read some posts that say it isn't as good as the previous versions. Should I install 1.2.8 or 1.2.7.1? I've had no issues at all with 1.2.9.1, however there have been several patches applied to the svn which I don't believe are part of the distro packages as yet. (My system is very basic with no need for queues, etc.) ___ --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
RE: [asterisk-users] Provider UNREACHABLE
Bill's right. But, it happens to me too, ALL the time, w/Teliax. I can't wait for their NYC node... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Wednesday, July 12, 2006 9:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Provider UNREACHABLE It's the internet...maybe for you the path to Teliax is kinda crappy? Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry Fawthrop Sent: Wednesday, July 12, 2006 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Provider UNREACHABLE Thanks All First off I never mentioned Teliax (but yes correctly ASSUMED they are my provider) and this is not a Teliax issue per se My issue is more the fact that I have Qualify = yes in sip.conf but repeatedly get REACHABLE and UNREACHABLE as can be seen below. even when I set Qualify = 3600 I still get this My question is more (a) how do I stop this ? (b) What is happening ? Thanks all Barry snip... Jul 11 13:10:38 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 57 Jul 11 13:10:48 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (117ms / 2000ms) Jul 11 13:18:01 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 13:24:17 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 2432 Jul 11 13:24:40 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (63ms / 2000ms) Jul 11 14:08:47 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 153 Jul 11 14:08:57 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (66ms / 2000ms) Jul 11 18:12:21 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 52 Jul 11 18:12:31 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (62ms / 2000ms) Jul 11 18:19:44 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:03:02 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:08:07 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 89 Jul 11 19:08:17 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:17:21 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 50 Jul 11 19:17:31 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 23:01:50 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 50 Jul 11 23:02:01 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) snip. ___ --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/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/asterisk-users
Re: [asterisk-users] Polycom, TFTP, and DHCP
Michael, Maybe I am not understanding your question, are you saying that when you configure your phone with a static IP address, you cannot find the boot server and when in DHCP you can? If you are having problems with the phone having a static IP address, make sure it is getting the correct IP, subnet, gateway and DNS. If your DNS is incorrect for example, you won't be able to find the server you entered, since there will be nothing to point the phone where to go. If you are talking about the actual boot server location, that needs to be static as far as I know. It isn't like DHCP addressing where it gets the DNS information from the host. It's a parameter that needs to be set. If your TFTP server is changing IPs I would strongly suggest giving it a static IP. It will make your life a lot easier. Kevin Michael Welter wrote: When I set the tftp address into the IP501 server parameters and boot, the phone says it says it cannot find the boot loader and reuses the previous configuration. When I set the tftp address in DHCP and reboot the phone, it finds the tftp server and loads correctly. My problem is that I don't always have control of the DHCP server. Is there a way to set the phone to find the tftp server on its own? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk + fax
Hi all I need a help with asterisk+fax - fax to email I am trying to setup fax to emailwith asterisk with nosuccess.I have asterisk 1.2.9.1 running on CentOS i have created extension 300 which should receive faxes.extensions.conf - exten = 300,1,Goto(fax,s,1)exten = 300,2,Congestionexten = 300,3,Hangupexten = s,1,Macro(faxreceive)exten = h,1,system(/usr/bin/mail -s "Fax from ${CALLERIDNUM} ${CALLERIDNAME}" ${EMAILADDR} ${FAXFILE})[macro-faxreceive]exten = s,1,Set(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)exten = s,2,Set(EMAILADDR=${FAX_RX_EMAIL})exten = s,3,rxfax(${FAXFILE}|debug)exten = s,103,Set(EMAILADDR=${FAX_RX_EMAIL})exten =
s,104,Goto(3) When i am trying to call 300 extension i am receiving broken fax noise. in addition on the CLI i see the next line Executing RxFAX("SIP/5060-08d6f170", "/var/spool/asterisk/fax/1152714504.466.tif|debug") in new stackBut the file never been created. In /var/log/asterisk/full i see these lines:Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier downJul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier upJul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier downJul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier upJul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier downJul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier upJul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier downJul 12
17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier upJul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier downJul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier upJul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier downJul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier upJul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier downJul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up Any help with fax to email with Asterix will be appreciated.
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Re: [asterisk-users] Global variables and AGI
Yes, thanks again for the suggestions. I wrote a few scripts for different things that we needed in the office and by the time I got to that one, I was tired and wasn't thinking straight anymore. I am probably going to just set a dummy variable for now and have asterisk update the global. Down the road we plan on adding a database for call logging, configurations, etc, and I would agree with you Jay, storing the variable there would be the better choice. Thanks again. Kevin Jay Milk wrote: Kevin Smith wrote: Hi everyone, I know that functions like set_variable and get_variable (using php with phpagi) only apply to the channel variable. What I need to do is reset a global variable I have in our system. I have a script that is going to determine when this will happen, but I just have to make it happen. Assuming that I cannot update the variable via the script, it is there a way I can make a call to the system, such as a call file, and place it in the context of the dialplan that I need to change the variable? If so, is there anything special I need in the call file for that to work? Or is there a easier/better way to do this that I haven't thought of. Any suggestions would be helpful. Thanks, Kevin As Timebandit pointed out -- http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Set or SetGlobalVar in 1.0.x If most of the interaction with that variable occurs through agi, you might also want to consider storing it outside of Asterisk. I've stored a good number of values in mysql for an asterisk application before. If most of the interaction occurs within the dialplan and/or you're trying to avoid agi, you could also use the asterisk database directly with DBPut and DBGet. ___ --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/asterisk-users
[asterisk-users] comcast info -- somewhat offtopic
A comcast representative told me the other day they are planning on doubling their internet speed from 8Mb to 16Mb at the end of this year. :-D signature.asc Description: OpenPGP digital signature ___ --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
Re: [asterisk-users] Problem with making outgoing calls
Crazy Boy wrote:
Hi,
We could
makecalls to USA using Teliax service upto 11, July 2006 with
Asterisk. But, since 11, July 2006 evening, we are unable to make calls
sometimes and could not connect to Teliax server sometimes. I have
realized that Teliax server was down for few hours.
Currently
our Asterisk server is connecting with Teliax. But, When I am trying to
make call to USA, Its giving me one ring and being disconnected. I
could not understand what could be the problem? Isthere any problem
with my connection to Teliax server?
With the
configuration mentioned below, we could successfully make calls to USA.
But, from this morning we are unable to make calls to USA.
IAX.CONF
file contents:
disallow=all
allow=ulaw
SIP.CONF file contents:
[101]
type=friend
username=101
secret=abcd
callerid="Ani"
host=dynamic
context=tutorial
[general]
register = ab.cd:[EMAIL PROTECTED]
[authentication]
auth = ab.cd:[EMAIL PROTECTED]
[teliax]
context=default
type=friend
username=ab.cd
user=ab.cd
host=voip-co1.teliax.com
secret=xx
insecure=very
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
EXTENSIONS.CONF
file contents:
exten
= 101,1,Dial(SIP/101,15)
exten
= 101,2,Voicemail(u101)
exten
= 101,3,Voicemail(b101)
exten
= 101,4,Hangup
exten =
_1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr)
exten = 3031234567,1,Answer()
exten = 3031234567,2,DIAL(SIP/user,20)
VOICEMAIL.CONF file
contents:
101 = ani, Ani, [EMAIL PROTECTED], [EMAIL PROTECTED]
Please
let me know the problem ASAP. Looking forward to your response.
Thank you.
Regards,
Chandra.
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Do you get some sort of error if you look at the console (CLI) while
attempting the call? If not, give it a try.. If you try and don't see
anything useful, try turning on debug by typing "sip debug" if using
sip or "iax2 debug" if using IAX.
When you are done use "sip no debug" or "iax2 no debug" to turn it off
:)
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Origination/Termination with SUPERIOR customer service!
http://www.VoIPstreet.com
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Re: Re: [Asterisk-Users] app_conference DTMFs?
Hello, My backtraces never actually mention play_sound, but the crashes only happen right after app_conference attempts to play out DTMF tines with the playing function. Here's the backtrace for two of the crashes that we had with app_conference: http://205.201.151.24/files/app_conference-crash-2006-06-02.txt http://205.201.151.24/files/app_conference-crash-2006-06-05.txt MATT--- On 7/12/06, jeff oconnell [EMAIL PROTECTED] wrote: interesting. i didn't realize the problem seems to specifically be the sound playback via the manager interface. a couple weeks ago, my asterisk on my dev box crashed, i did some preliminary investigation, but since we hadn't had any problems in production or qa, i chalked it up to me messing up my dev install somehow. anyway, here's the backtrace of the core: (gdb) bt #0 0x00df71bb in ?? () from /lib/libgcc_s.so.1 #1 0x080625ad in ast_deactivate_generator (chan=0x815fb40) at channel.c:1382 #2 0x0806d77a in ast_openstream_full (chan=0x815fb40, filename=0x815dcec tones-exit, preflang=0x0, asis=0) at file.c:494 #3 0x0806d835 in ast_openstream (chan=0x71bfb8a6, filename=0x71bfb8a6 Address 0x71bfb8a6 out of bounds, preflang=0x71bfb8a6 Address 0x71bfb8a6 out of bounds) at file.c:467 #4 0xb7e8899b in conference_play_sound (fd=12, argc=14643636, argv=0xdfdc5c) at cli.c:225 #5 0x0809786b in ast_cli_command (fd=12, s=0x71bfb8a6 Address 0x71bfb8a6 out of bounds) at cli.c:1364 #6 0x080b2f8a in action_command (s=0x815c448, m=0xb7b97420) at manager.c:927 #7 0x080b7b81 in process_message (s=0x815c448, m=0xb7b97420) at manager.c:1305 #8 0x080b83cf in session_do (data=0x815c448) at manager.c:1401 #9 0x00b44341 in start_thread () from /lib/tls/libpthread.so.0 #10 0x009096fe in clone () from /lib/tls/libc.so.6 the interesting lines to me are #4 and #5: #4 0xb7e8899b in conference_play_sound (fd=12, argc=14643636, argv=0xdfdc5c) at cli.c:225 #5 0x0809786b in ast_cli_command (fd=12, s=0x71bfb8a6 Address 0x71bfb8a6 out of bounds) at cli.c:1364 line 4 becauset the argc passed into conference_play_sound() is so large, and line 5 because there seems to be an out--of-bounds problem in the asterisk code ( i.e. before the app_conference code is called ). based on what you said in your last post, i'm going to look at this more. if you have any thoughts on my backtrace/analysis, let me know. j- On 7/12/06, Matt Florell [EMAIL PROTECTED] wrote: The issues with the double-free crashes that we've had all seem to be caused by playing of audio files(like the entry/exit sounds or the DTMF broadcast). But these functions use the app_conference code that was already existing to play audio files from manager API commands so the issue was there it just wasn't as tested because not many people use the manager command a lot to play audio in conferences. ___ --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
Re: [asterisk-users] NuFone, please send the log file
Kevin P. Fleming wrote: Can we please keep the discussions about carriers, money, jobs, work, etc. off of this list? This is not the place to discuss your experiences with _any_ company, it's a place to talk about Asterisk and using Asterisk. Please move flamewars and similar discussions to some other forum. I agree with you! Which place is in your opinion the right place? As long there is no other place, such messages will always pop up. bye Ronald ___ --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
Re: [asterisk-users] New Asterisk server crashes daily
On Tue, 2006-07-11 at 14:39 +0100, Roshan Sembacuttiaratchy wrote: On Tue, Jul 11, 2006 at 06:23:05AM -0700, Al Lougher scribbled: Hi - This is the first Linux server I have ever built with an installation of [EMAIL PROTECTED] 2.7. For development I have been running on VMWare on an XP box and sustained no crashes or reboots. After moving Asterisk to it's own server I am experiencing daily crashses (around 4am) and I'm not quite sure what the problem is, nor am I sure where exactly to look for logs of any errors prior and during the crash. During the crash there should be nothing running so I'm not sure why it crashes at this time (perhaps some system job that is running at this time?). My hardware is: AMD Athlon 64bit 3200 CPU, 1 gig memory, 100gb hd and a gigabit NIC card. The BIOS is set with defaults. Many thanks, Al. Your comment about it happening around 4am leads me to think it might be the default daily-scheduled cron jobs somehow affecting you. Are you sure you have enough swap space configured? Try disabling prelink in cron. Regards, Patrick ___ --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
Re: [asterisk-users] Asterisk + fax
On 7/12/06, al gav [EMAIL PROTECTED] wrote: Hi all I need a help with asterisk+fax - fax to email I am trying to setup fax to email with asterisk with no success. I have asterisk 1.2.9.1 running on CentOS i have created extension 300 which should receive faxes. Which version of spandsp did you build/install? What type of fax machine are you using to send with, and have you tried a different one? Steve ___ --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
[asterisk-users] Email notification of voicemail
Asterisk is trying to send an email to users when they receive a voicemail. Can this be shut off? I have not entered any email addresses in voicemail.conf so it tries to send to [EMAIL PROTECTED]. This of course gets rejected since the user does not exist and the root users mailbox on linux gets full of these rejection notices. I cant seem to find anywhere to tell Asterisk to stop notifying people they have voicemails. Im using 1.2.9.1 of Asterisk. Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, 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
Re: [asterisk-users] comcast info -- somewhat offtopic
and cable vision now has 30/2 and they will have 50/50 real soon - Original Message - From: Derek Whitten [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 12, 2006 10:45 AM Subject: [asterisk-users] comcast info -- somewhat offtopic ___ --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/asterisk-users
Re: [asterisk-users] Asterisk + fax
Hi all
I need a help with asterisk+fax - fax to email
I am trying to setup fax to email with asterisk with no success.
I have asterisk 1.2.9.1 running on CentOS
i have created extension 300 which should receive faxes.
extensions.conf
-
exten = 300,1,Goto(fax,s,1)
exten = 300,2,Congestion
exten = 300,3,Hangup
exten = s,1,Macro(faxreceive)
exten = h,1,system(/usr/bin/mail -s Fax from ${CALLERIDNUM}
${CALLERIDNAME} ${EMAILADDR} ${FAXFILE})
[macro-faxreceive]
exten = s,1,Set(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,Set(EMAILADDR=${FAX_RX_EMAIL})
exten = s,3,rxfax(${FAXFILE}|debug)
exten = s,103,Set(EMAILADDR=${FAX_RX_EMAIL})
exten = s,104,Goto(3)
When i am trying to call 300 extension i am receiving broken fax noise.
in addition on the CLI i see the next line
Executing RxFAX(SIP/5060-08d6f170,
/var/spool/asterisk/fax/1152714504.466.tif|debug) in new stack
But the file never been created.
In /var/log/asterisk/full i see these lines:
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up
Any help with fax to email with Asterix will be appreciated.
-
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The echo cancellation on your line is causing this. I had the same issue
with faxing.
Garth
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Re: [asterisk-users] NuFone, please send the log file
trixter aka Bret McDanel wrote: On Tue, 2006-07-11 at 20:51 -0400, C F wrote: While I don't disagree with you, look at what my point was, just accusing them for such without any documentation doesn't make sens. He only brought that up after people started questioning it. So I dunno. And lets face it, this is the internet there is really no proof of anything. Screen captures of a webpage? That is easy enough to forge. Invoices? They too are easy enough to forge. I don't think so!!! I guess you never lost your web site (accidentally) a have been than very happy that at least a big portion you could retrieve from the Internet archive!!! It is even funny to see how some web pages have been developed and changed. Even if someone states they had horrible call quality you have no proof, but that is generally accepted that that one person experienced that. And where does that leave you? You have to either take a chance on your own or go with those that you trust and/or whatever is said the most. Call quality changes often and in my experience depends not so often from the VoIP provider, but from the users Internet connection. bye Ronald So since its hard to get any sort of proof you kinda just have to accept that it happened or not and move on. ___ --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 --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0628-2, 2006/07/11 Tested on: 2006/7/12 ¤W¤È 09:27:28 avast! - copyright (c) 1988-2006 ALWIL Software. http://www.avast.com -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ --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
Re: Re: [Asterisk-Users] app_conference DTMFs?
Matt Florell [EMAIL PROTECTED] wrote: My backtraces never actually mention play_sound, but the crashes only happen right after app_conference attempts to play out DTMF tines with the playing function. This is because Malloc isn't crashing when the mistake is made. It crashes later because of the out of bounds write or double free has corrupted its memory structures. -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --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
Re: [asterisk-users] comcast info -- somewhat offtopic
Dovid Bender wrote: and cable vision now has 30/2 and they will have 50/50 real soon - Original Message - From: Derek Whitten [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 12, 2006 10:45 AM Subject: [asterisk-users] comcast info -- somewhat offtopic ___ --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/asterisk-users nice! signature.asc Description: OpenPGP digital signature ___ --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
Re: [asterisk-users] Asterisk + fax
On 7/12/06, al gav [EMAIL PROTECTED] wrote:
Hi all
I need a help with asterisk+fax - fax to email
I am trying to setup fax to email with asterisk with no success.
I have asterisk 1.2.9.1 running on CentOS
i have created extension 300 which should receive faxes.
extensions.conf
-
exten = 300,1,Goto(fax,s,1)
exten = 300,2,Congestion
exten = 300,3,Hangup
exten = s,1,Macro(faxreceive)
exten = h,1,system(/usr/bin/mail -s Fax from ${CALLERIDNUM}
${CALLERIDNAME} ${EMAILADDR} ${FAXFILE})
[macro-faxreceive]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,Set(EMAILADDR=${FAX_RX_EMAIL})
exten = s,3,rxfax(${FAXFILE}|debug)
exten = s,103,Set(EMAILADDR=${FAX_RX_EMAIL})
exten = s,104,Goto(3)
When i am trying to call 300 extension i am receiving broken fax noise.
in addition on the CLI i see the next line
Executing RxFAX(SIP/5060-08d6f170,
/var/spool/asterisk/fax/1152714504.466.tif|debug) in new
stack
But the file never been created.
In /var/log/asterisk/full i see these lines:
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier down
Jul 12 17:28:24 DEBUG[25029] app_rxfax.c: FLOW HDLC carrier up
Any help with fax to email with Asterix will be appreciated.
Do you Yahoo!?
Everyone is raving about the all-new Yahoo! Mail Beta.
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Well, the fact that you are calling the extension and hear the fax
signaling shows the that rxfax application is working, so far so good.
Have you set faxdetect=incoming in /etc/asterisk/zapata.conf ?
Also note that you need to have an extension named fax in your
default context.
Also look at http://www.voip-info.org/wiki-Asterisk+fax for helping info.
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[asterisk-users] where the bottleneck lies ? (was: Server redundancy)
unplug ha scritto: I feel interested about you can support 16,000 users of your system. As I have tested using sipp in a dual CPU Xeon with 2G Ram, the maximum number of current call is about 160. In some forums, most of ppl claim the maximum current call is about 100-200. What do you expect the number of current call to handle in 16,000 users? I'm curious about what was limiting the number of calls in your tests. For every system I have in production/testing I see the only bottleneck is system load, cpu and memory usage is well beyond limits when things starts to fall apart. The unexplicable (at least by me) thing is that system load seems to be only partially influenced by the number of calls, for example sometimes there are 100/150 calls and the load is around 0.70, sometimes it skyrockets to 2.00 / 2.50 (when it is 2 calls quality is crippled, I think because of too many dropped packets). I see this behaviour no matter how simple/complex the system is, from just a terminator with a couple of digium in it and a five-lines extension to the central server with fastagi doing mysql queries and taking hundreds of concurrent calls in both sip and iax. Can it be something related to asterisk itself ? I'm thinking about installing oprofile on the various servers, someone by chance already did it ? ___ --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
[asterisk-users] Call Parking breaks suddenly
Hi, We're using Polycom IP501 SIP phones (app version 1.6.4.0043) with Asterisk 1.2.9.1. I set up call parking last week and for a while, it worked great. It stopped working yesterday, all of the sudden. What happens is that when the phone user dials #999 (our parkext), the call does not get parked and the caller hears the DTMF. Actually, they don't hear the DTMF, they hear a popping noise as the keys are pressed. The configuration files have not been changed since call parking was initially enabled. I'm running a console with -vvv and I don't see any errors reported. Any ideas? Thanks... Chris ___ --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
Re: [asterisk-users] 1000s of extensions in one context?
Dovid Bender schrieb: several thousand extensions or several extensions called 1000 ? Several thousend extensions. exten = 497111234,1,goto(...) exten = 497111235X,1,goto(...) exten = 497111236XX,1,goto(...) exten = 497111237,1,goto(...) Several thousend extensions of maybe different length. For overlap dialing to operate correct (and no need to wait for timeouts) I would like to put the whole dial plan into the file extensions.conf. Before starting, I would like to know, whether there are experiences with such long dialplans. Roger. ___ --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
Re: [asterisk-users] Asterisk version: 1.2.9.1 or older?
On Jul 12, 2006, at 7:16 AM, Rich Adamson wrote: Andrea Spadaccini wrote: Hello, I need to install Asterisk on a test machine that will soon become a production environment. Do you think that 1.2.9.1 is reliable? I read some posts that say it isn't as good as the previous versions. Should I install 1.2.8 or 1.2.7.1? I've had no issues at all with 1.2.9.1, however there have been several patches applied to the svn which I don't believe are part of the distro packages as yet. (My system is very basic with no need for queues, etc.) Same here, about three weeks of uptime on 1.2.9.1 with no issues, but a very simple setup. YMMV. ___ --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
Re: [asterisk-users] Provider UNREACHABLE
On Jul 12, 2006, at 7:24 AM, Rick Smith wrote: Bill's right. But, it happens to me too, ALL the time, w/Teliax. I can't wait for their NYC node... I found that sellvoip.net is closer to me here in Seattle and also has a better rate. They also have a server in NYC. I still like Teliax and use them as my primary backup with Nufone being the third configured terminator. Sellvoip.net hasn't been perfect (ie there have been some short outages) but the route from me to them is so short that my call quality issues disappeared instantly (or in 15ms). Marty ___ --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
Re: [asterisk-users] Email notification of voicemail
Kevin Savoy wrote: Asterisk is trying to send an email to users when they receive a voicemail. Can this be shut off? I have not entered any email addresses in voicemail.conf so it tries to send to [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]. This of course gets rejected since the user does not exist and the root users mailbox on linux gets full of these rejection notices. I can’t seem to find anywhere to tell Asterisk to stop notifying people they have voicemails. I’m using 1.2.9.1 of Asterisk. Thanks _ **Kevin Savoy** **Business Unit Telecom Analyst** 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, 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 You could try commenting out: attach=yes Also, if you don't want any emails sent ever for any voice mail users you could probably uncomment the following line and give it a bogus path to the mailer. ;mailcmd=/usr/sbin/sendmail -t There is probably a better way to do this but we have never needed to turn it off so I am not sure. Hope this helps. -- VoIP Street Origination/Termination with SUPERIOR customer service! http://www.VoIPstreet.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
Re: [asterisk-users] comcast info -- somewhat offtopic
On Jul 12, 2006, at 7:45 AM, Derek Whitten wrote: A comcast representative told me the other day they are planning on doubling their internet speed from 8Mb to 16Mb at the end of this year. They certainly don't deliver anywhere near 8Mbits per second here... So I don't know what those kind of promises mean. I had about 4 times the bandwidth when it was an @home connection. All down hill since. Marty ___ --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
[asterisk-users] PCMCIA card support
Hi all,I'm new to asterisk and I just installed it.I already have a PCMCIA THOR-2 which supports two T1/E1 and I noticed asterisk doesn't support this card, but I fount over the ineternet that I would have to write a glue code in order to work with this card. I'm not familiar with this code writing, but I'd really appreciate if someone could give me a clue on where to start, or if anyone has done something similiar before.I'm looking forward to work with this card. Thanks in advance,Mauricio ___ --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
[asterisk-users] Exclude a certain route from using a trunk
Hi,In my Outbound routes i have created International Local Calls. I have 2 trunks for both ITL and LC. All calls are dialed using 011.but all 011254, 01125473, 01125472 should use the local trunk. NB Local Route is 1st priority in my list or routes. Everyone has to dial 011(number) to make a call whether Local or international but all 011254* number should use my local trunk. How do i achieve this? This is what i have so far; Outbound Route - (International Calls) **Ive put the same in the trunksDial Pattern 011.Outbound Route - (Local Calls) **Ive put the same in the trunksDial Pattern 011254|072XXX011254|073XXX 011254|K ___ --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
RE: [asterisk-users] Provider UNREACHABLE
I have the same problem before with 2 different providers. We resolved this by turning off qualify (qualify=no). KC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Filip Dragowski Sent: Wednesday, July 12, 2006 6:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Provider UNREACHABLE (b) What is happening ? If i unplug network cable from my ipphone asterisk will say UNREACHABLE after few seconds. Sometimes it occurs for my sip provider. I simply lose connection with my phone/provider or connection is so poor that transmission of voice will be very choppy, so beeter not to start a call at all. Mayby someone download big files in Your network and take all bandwitch ? for me it's a network issue. -Filip Użytkownik Barry Fawthrop napisał: Thanks All First off I never mentioned Teliax (but yes correctly ASSUMED they are my provider) and this is not a Teliax issue per se My issue is more the fact that I have Qualify = yes in sip.conf but repeatedly get REACHABLE and UNREACHABLE as can be seen below. even when I set Qualify = 3600 I still get this My question is more (a) how do I stop this ? (b) What is happening ? Thanks all Barry snip... Jul 11 13:10:38 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 57 Jul 11 13:10:48 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (117ms / 2000ms) Jul 11 13:18:01 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 13:24:17 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 2432 Jul 11 13:24:40 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (63ms / 2000ms) Jul 11 14:08:47 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 153 Jul 11 14:08:57 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (66ms / 2000ms) Jul 11 18:12:21 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 52 Jul 11 18:12:31 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (62ms / 2000ms) Jul 11 18:19:44 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:03:02 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:08:07 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 89 Jul 11 19:08:17 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 19:17:21 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 50 Jul 11 19:17:31 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) Jul 11 23:01:50 NOTICE[381] chan_sip.c: Peer 'teliax' is now UNREACHABLE! Last qualify: 50 Jul 11 23:02:01 NOTICE[381] chan_sip.c: Peer 'teliax' is now REACHABLE! (50ms / 2000ms) snip. ___ --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/asterisk-users
RE: [asterisk-users] where the bottleneck lies ? (was: Serverredundancy)
-Original Message- From: Simone Cittadini [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 12, 2006 10:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] where the bottleneck lies ? (was: Serverredundancy) unplug ha scritto: I feel interested about you can support 16,000 users of your system. As I have tested using sipp in a dual CPU Xeon with 2G Ram, the maximum number of current call is about 160. In some forums, most of ppl claim the maximum current call is about 100-200. What do you expect the number of current call to handle in 16,000 users? I'm curious about what was limiting the number of calls in your tests. For every system I have in production/testing I see the only bottleneck is system load, cpu and memory usage is well beyond limits when things starts to fall apart. The unexplicable (at least by me) thing is that system load seems to be only partially influenced by the number of calls, for example sometimes there are 100/150 calls and the load is around 0.70, sometimes it skyrockets to 2.00 / 2.50 (when it is 2 calls quality is crippled, I think because of too many dropped packets). I see this behaviour no matter how simple/complex the system is, from just a terminator with a couple of digium in it and a five-lines extension to the central server with fastagi doing mysql queries and taking hundreds of concurrent calls in both sip and iax. Can it be something related to asterisk itself ? I'm thinking about installing oprofile on the various servers, someone by chance already did it ? Another consideration is if the phones have performed reinvites, and removed Asterisk from the RTP stream. If you can live without call recording, and other features where Asterisk has to remain in the RTP path, then I imagine that this would significanlty reduce load on the Asterisk systems. Could some of your phones be reinviting? This may explain the variation in load. Doug. ___ --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
Re: [Asterisk-Users] dtmfmode=inband but SDP also indicates rfc2833
Indicating rfc2833 is exactly what asterisk does when it receives an invite from a server or device that indicates rfc2833 is available regardless of whether or not dtmfmode=inband. I will get a sip debug and open a bug report when I have a few minutes. Kevin P. Fleming wrote: - Stagg Shelton [EMAIL PROTECTED] wrote: I did ultimately force asterisk to the point where it will not accept or send rfc2833. I did this by modifying chan_sip.c in the function Asterisk should not be sending an SDP with RFC-2833 in it when the dtmfmode=inband in sip.conf. If it is doing that, please capture a 'sip debug' of this happening and opening a bug on bugs.digium.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
Re: [asterisk-users] comcast info -- somewhat offtopic
Martin Joseph wrote: On Jul 12, 2006, at 7:45 AM, Derek Whitten wrote: A comcast representative told me the other day they are planning on doubling their internet speed from 8Mb to 16Mb at the end of this year. They certainly don't deliver anywhere near 8Mbits per second here... So I don't know what those kind of promises mean. I had about 4 times the bandwidth when it was an @home connection. All down hill since. Marty ___ --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 i just upgraded to 8M and my avg d/l speed went up to between 850KB/s - 1.05MB/s signature.asc Description: OpenPGP digital signature ___ --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
[asterisk-users] an operational scenario
I'm trying to do something I've not see written up here before. I have an asterisk on a box with 2 interfaces like the drawing below. I want to have SIP extensions regsitering to both interfaces and able to communicate. Is this possible? What suggestions do you have? +-+ | | internal | | external --+ +- 192.168.1 | | real IP | | +-+ ___ --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
[asterisk-users] FXS adapters and Polycom phones
Hi, I`m looking for a SIP-PSTN adapter, basically to switch a customer from a cheap PBX to mine, but resuing their own Norstar PSTN phones. They have 10phones. >From a price point of view, it seems that 10 individual GrandStream SIP adapters is the best way to go, but it seems so inelegant to me. What is recommended ? Second question: I have a GrandStream GXP-2000, that despite what everybody says I love. I am still looking for a replacement, if only because it doesn`t look as good and it does have a few quirks. I was looking at Polycoms, but some questions are unanswered by looking at their datasheet. - Does the Polycom 501 have an integrated router (like the GXP-2000, latest firmware, does) - Can you have more than one SIP/account on the phone, each ringing in a way that lets the user know which account is ringing? (GXP2000 does it by making it possible to have each "line" linked to a separate SIP account) Thank you, Mike ___ --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
Re: [asterisk-users] Polycom, TFTP, and DHCP
Kevin Smith wrote: Michael, Maybe I am not understanding your question, are you saying that when you configure your phone with a static IP address, you cannot find the boot server and when in DHCP you can? The phone uses DHCP to get its IP address. In the phone's server params, I enter the IP address of the tftp server. Without the next-server entry in the DHCP configs, the phone says it cannot find the boot server (and uses the previous configuration). However, when next-server in DHCP is set with the tftp IP, the phone loads its configuration from tftp and boots normally. I'd like to not have to set the tftp address in DHCP, because I don't always have access to the DHCP server. Is there someway to tell the phone to override the DHCP server setting? Is there something I'm missing with the phone's network config? Thanks If you are having problems with the phone having a static IP address, make sure it is getting the correct IP, subnet, gateway and DNS. If your DNS is incorrect for example, you won't be able to find the server you entered, since there will be nothing to point the phone where to go. If you are talking about the actual boot server location, that needs to be static as far as I know. It isn't like DHCP addressing where it gets the DNS information from the host. It's a parameter that needs to be set. If your TFTP server is changing IPs I would strongly suggest giving it a static IP. It will make your life a lot easier. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --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
Re: [asterisk-users] Provider UNREACHABLE
On Jul 12, 2006, at 10:18 AM, KC wrote: I have the same problem before with 2 different providers. We resolved this by turning off qualify (qualify=no). Of course this doesn't fix anything, it just stops the warnings from showing up... Marty ___ --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
Re: [asterisk-users] 1000s of extensions in one context?
My working experience with 100s of extensions, usually associated to
personnel that will *not* change from my defaults is:
; Extensions
exten = 1000,1,Macro(call-sip-local,1000,SIP/1000,default) ; Operator
exten = _1XXX,1,Macro(call-sip-local,${EXTEN},SIP/${EXTEN},default)
Then,
[macro-call-sip-local]
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
; ${ARG3} - voicemailcontext
;
exten = s,1,Set(LANGUAGE()=en)
exten = s,n,Playback(pls-wait-connect-call)
exten = s,n,Set(LANGUAGE()=es)
exten = s,n,Dial(${ARG2},20,tT) ; Ring the
interface, 20 seconds maximum
exten = s,n,NoOp(${DIALSTATUS})
exten = s,n,Goto(s-${DIALSTATUS},1) ; Jump based on
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION$
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED]) ; If
unavailable, send to voicemail w/ unavail announce
exten = s-NOANSWER,n,HangUp()
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED]) ; If
busy, send to voicemail w/ busy announce
exten = s-BUSY,n,HangUp()
exten = s-CHANUNAVAIL,1,PlayTones(congestion)
exten = s-CHANUNAVAIL,n,Wait(2)
exten = s-CHANUNAVAIL,n,StopPlayTones()
exten = s-CHANUNAVAIL,n,Voicemail([EMAIL PROTECTED])
exten = s-CHANUNAVAIL,n,HangUp()
exten = _s-.,1,Goto(s-NOANSWER,1) ;
Treat anything else as no answer
;
; end;
On 7/12/06, Roger Schreiter [EMAIL PROTECTED] wrote:
Dovid Bender schrieb:
several thousand extensions or several extensions called 1000 ?
Several thousend extensions.
exten = 497111234,1,goto(...)
exten = 497111235X,1,goto(...)
exten = 497111236XX,1,goto(...)
exten = 497111237,1,goto(...)
Several thousend extensions of maybe different length.
For overlap dialing to operate correct (and no need to
wait for timeouts) I would like to put the whole dial
plan into the file extensions.conf.
Before starting, I would like to know, whether there are
experiences with such long dialplans.
Roger.
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--
Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780
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Re: [Asterisk-Users] app_conference DTMFs?
Henry J. Cobb wrote: I tried several different combinations of app_conference and Asterisk versions and then I had to get back to actually providing phone service that didn't crash. I hate to me-too, but my experience was identical. Crash after crash, and I tried everything that was suggested (limiting codecs, primarily). Something is weird there in that for some it appears to work perfectly, for others not at all. . . FWIW. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
Re: [asterisk-users] an operational scenario
Why can't you do it? I have an internal 192.168.100.x address (eth0) and 200.x.x.x (eth1) interface. Internal users register to the 192 and internet users register to the 200.x address internal extensions are 1XXX and external extensions are 2XXX What errors do you have? On 7/12/06, Bruce Ferrell [EMAIL PROTECTED] wrote: I'm trying to do something I've not see written up here before. I have an asterisk on a box with 2 interfaces like the drawing below. I want to have SIP extensions regsitering to both interfaces and able to communicate. Is this possible? What suggestions do you have? +-+ | | internal | | external --+ +- 192.168.1 | | real IP | | +-+ ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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
RE: [asterisk-users] Email notification of voicemail
I have attach=no in my voicemail.conf so that can't be doing it. Not sure where that sendmail command is. Don't see it in voicemail.conf or any other config in the asterisk directory. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of VoIP Street Sent: Wednesday, July 12, 2006 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Email notification of voicemail Kevin Savoy wrote: Asterisk is trying to send an email to users when they receive a voicemail. Can this be shut off? I have not entered any email addresses in voicemail.conf so it tries to send to [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]. This of course gets rejected since the user does not exist and the root users mailbox on linux gets full of these rejection notices. I can't seem to find anywhere to tell Asterisk to stop notifying people they have voicemails. I'm using 1.2.9.1 of Asterisk. Thanks _ **Kevin Savoy** **Business Unit Telecom Analyst** 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, 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 You could try commenting out: attach=yes Also, if you don't want any emails sent ever for any voice mail users you could probably uncomment the following line and give it a bogus path to the mailer. ;mailcmd=/usr/sbin/sendmail -t There is probably a better way to do this but we have never needed to turn it off so I am not sure. Hope this helps. -- VoIP Street Origination/Termination with SUPERIOR customer service! http://www.VoIPstreet.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 or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] an operational scenario
the problem I'm seeing is one way audio between extensions. I've splpit up the numbering plan internal/external. All are in the same range. I'll try splitting them and see what happens. Erick Perez wrote: Why can't you do it? I have an internal 192.168.100.x address (eth0) and 200.x.x.x (eth1) interface. Internal users register to the 192 and internet users register to the 200.x address internal extensions are 1XXX and external extensions are 2XXX What errors do you have? On 7/12/06, Bruce Ferrell [EMAIL PROTECTED] wrote: I'm trying to do something I've not see written up here before. I have an asterisk on a box with 2 interfaces like the drawing below. I want to have SIP extensions regsitering to both interfaces and able to communicate. Is this possible? What suggestions do you have? +-+ | | internal | | external --+ +- 192.168.1 | | real IP | | +-+ ___ --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 -- One day at a time, one second if that's what it takes ___ --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
Re: [Asterisk-Users] app_conference DTMFs?
thanks brian, this is all really helpful feedback!just to be clear, which app_conference code were you using?the svn trunk version from sourceforge? or the VD_app_conference matt's been working on? j-On 7/12/06, Brian Capouch [EMAIL PROTECTED] wrote: I hate to me-too, but my experience was identical.Crash after crash,and I tried everything that was suggested (limiting codecs, primarily).Something is weird there in that for some it appears to work perfectly, for others not at all. . . ___ --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
