Re: [asterisk-users] Voicemail, Asterisk and Grandstream BT200
Make sure you have busydetect=yes busycount=3 somewhere below your [general] context in chan_dahdi.conf (or zapata.conf depending on your asterisk version) and restart the the service. This should be enoough to do the magic. Alyed 2010/3/21 Daniel Bareiro daniel-lis...@gmx.net -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Gordon. On Sun, 21 Mar 2010, Gordon Henderson wrote: I'm testing with a Grandstream BT200 telephone and, according to I read, it has a LED that blinks if for that extension messages were left. In Voice Mail UserID, under the ACCOUNT tab, I put *100 that is the extension in which my Asterisk answer the voicemail service and if then I press MESSAGE button, the telephone communicates with Asterisk and, after to introduce the password, it indicates to me that I have messages. But the luminous indicator does not work. It is necessary to configure something special for this? It can be that it doesn't work because there is to introduce one password previously? There's another setting in the phone you need to set SUBSCRIBE for MWI. Yes. I was needing to indicate the use of MWI of the side of the configuration of the telephone. I selected the SUBSCRIBES for MWI checkbox. And make-sure the mailbox number is listed in the sip.conf entry for that phone. According to which I was reading, the MWI notifications become by the option mailbox= in the configuration of the extension. For this extension, the 104, had mailbox=104 but still with MWI enabled option, it was not working. After to think enough on this subject, I have noticed that instead of 104 I had to put 1...@voicemail since voicemail it was context that I'm using in voicemail.conf. With this already was working. However, beyond this, I was with the following situation: if I call from a cell phone, my Asterisk take the call, it presents to the caller the possibility to dialing an extension number and, in case of not doing it, it transfers this call to a specific extension. Then, if in this extension nobody takes the call, the service of voicemail is triggered so that the caller leaves its message from the cell phone. But if it hangs after to let the message without have pressed previously the pound key, the channel is taken and no longer any other call enters the PBX from the PSTN. This does not happen if the caller presses the pound key after to have left his message. As I have a box at which the cable arrives from the PSTN in which there are two ports of derivation and in one of them it leaves the cable for the Asterisk PBX (connected only then), after to have detected this problem I tried connecting in the other port an analog telephone and, indeed, it did not have tone as if never it had been hung. In addition this was confirmed because the MWI light never blinked on the telephone. After restarting the Asterisk server, yes the MWI light blinks and in addition I could corob the time in which the channel was taken seeing that the message lasted more than nine minutes. To what this problem can be due? It has to do the call is made specifically from cell phone through the PSTN (because if I leave a message hanging directly without pressing the pound key from an local extension, this does not happen)? There is some form to avoid it? Thanks for your reply! Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkums0oACgkQZpa/GxTmHTcGpQCghJvfphxc5ZzZhouryA+OlwGm 20AAoJP64a2EVeigx08D/5g5XN8oBXgf =Hskd -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Do i really need Dahdi and Libpri.
On Sun, 2010-03-21 at 19:00 -0400, Zeeshan Zakaria wrote: Good to know. I'll try that. I needed such solution for a client few months ago. On 2010-03-21 6:06 PM, Gordon Henderson gordon +aster...@drogon.net wrote: On Sun, 21 Mar 2010, Zeeshan Zakaria wrote: Virtual machine will not be able to access dahdi hard... It will depend on the virtualisation technology you use... I'm currently using LXC and can easily give a container full hardware access if required. Certianly no issues with dahdi_dummy... Gordon With XEN you can do forwarding of complete pci-devices. They become hidden from the main-system (dom-0) and exclusively available for the client (dom-U) Personally I've done it for crypto devices, I've seen on several lists that others have done it for network-, VGA-, isdn-, videocapture-boards. So i presume dahdi-boards will pose no exception... hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail, Asterisk and Grandstream BT200
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Alyed. On Mon, 22 Mar 2010, Alyed wrote: I was with the following situation: if I call from a cell phone, my Asterisk take the call, it presents to the caller the possibility to dialing an extension number and, in case of not doing it, it transfers this call to a specific extension. Then, if in this extension nobody takes the call, the service of voicemail is triggered so that the caller leaves its message from the cell phone. But if it hangs after to let the message without have pressed previously the pound key, the channel is taken and no longer any other call enters the PBX from the PSTN. This does not happen if the caller presses the pound key after to have left his message. As I have a box at which the cable arrives from the PSTN in which there are two ports of derivation and in one of them it leaves the cable for the Asterisk PBX (connected only then), after to have detected this problem I tried connecting in the other port an analog telephone and, indeed, it did not have tone as if never it had been hung. In addition this was confirmed because the MWI light never blinked on the telephone. After restarting the Asterisk server, yes the MWI light blinks and in addition I could corob the time in which the channel was taken seeing that the message lasted more than nine minutes. To what this problem can be due? It has to do the call is made specifically from cell phone through the PSTN (because if I leave a message hanging directly without pressing the pound key from an local extension, this does not happen)? There is some form to avoid it? Make sure you have busydetect=yes busycount=3 somewhere below your [general] context in chan_dahdi.conf (or zapata.conf depending on your asterisk version) and restart the the service. This should be enoough to do the magic. It didn't have configured these two parameters so I added now them but in the [channels] context since I don't have a [general] context (It does not sound to me that in the file by default generated by Asterisk there would not be it either, although I can be mistaken). Beyond that, with these two parameters, I no longer have the problem mentioned before. Thanks! However, the following doubt arises to me: it would also have had this problem for some originating call from a telephone that is not a cell phone? Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkunNjQACgkQZpa/GxTmHTfAbACfT8PVkcp/xESdqsiczg3YY/Dd FGcAn1TdOqiZaKAjLg4h3SDt/34A4bKX =37qZ -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Invalid Makefiles to install asterisk with ldap
On Sun, 2010-03-21 at 21:57 +0100, mickael wrote: I have a problem to install asterisk with ldap. . / configure That one should have found libldap make menuselect LIBS =- lldap make This is where my error .. / usr / bin / ld: can not find-lldap Do you have an /usr/lib/libldap.so ? Maybe part of openldap-devel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Thursday, March 18, 2010 1:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning I know for a fact that you can provision a Polycom via ftp. I've included much of my dhcpd.conf file below. Pick out what you need. Let me know if I can confirm that using option 66 will work with FTP (and HTTP, for that matter) with newer BootROM versions. I don't know the exact version it changed, unfortunately, as I just noticed it in passing when I was running some tests one day. As for why we ended up choosing option 129 originally rather than 160? I wish I had a clever technical explation, but it's just a random unassigned option number. That's it. :) - Brad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI lines do not have CallerID activated yet it is
Hi, I am having some trouble setting up Caller id on my asterisk system, I need to know if there is anything special that needs to be done for an australian connection specifically as I have tried what most web sites on google reccomend but without success. I have not had much experience with asterisk as I have inherited this system from the previous sysadmin who has not documented anything so I am unsure what data you guys need from me, please advise what is needed and I will get it to you asap. Thanks in advance -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemail problem
Hi people! I am running Asterisk 1.6.1.12 and I set up the voicemail. That far I have set upt the voicemailbox with my personal greeting message. If somebody calls me and is forwarded to my mailbox, my personal recorded greeting is played back + the default message please record your message after the tone and hang up or press the pound key. Is there a way to delete the second part from the voicemail, that only my personal recorded message is played back and a signal tone comes to signal the caller to start talking?! Tamer Higazi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI lines do not have CallerID activated yet it is
Exactly what is the problem you've having with CallerID? Are you not receiving it, or are you not able to send it? Which carrier are you using and what make and model card is the line connected to? For incoming calls on ISDN-10/20/30 lines, no special configuration is required to receive caller ID, however your carrier must be presenting it to you. Likewise no special configuration (beyond dialplan functionality to set the number to be sent) is required to set caller ID on outgoing calls, again provided your carrier will allow this. The format of the number to be set varies a great deal, but as a rule of thumb, you set the caller ID in the same format you receive it in. Any outgoing caller ID that does not fall within the allocated number range for the service will just about always be ignored, resulting in the default caller ID being presented to the called party. On 03/22/10 22:25, Nathanial Allan wrote: Hi, I am having some trouble setting up Caller id on my asterisk system, I need to know if there is anything special that needs to be done for an australian connection specifically as I have tried what most web sites on google reccomend but without success. I have not had much experience with asterisk as I have inherited this system from the previous sysadmin who has not documented anything so I am unsure what data you guys need from me, please advise what is needed and I will get it to you asap. Thanks in advance -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail problem
Since the application just does a playback of the canned sounds in /var/lib/sounds/asterisk, you can use SOX, Audacity, etc. to mix and chop these sounds in whatever way you see fit. Do a core set verbose 10 on the CLI and watch the output as you leave a voicemail to see which files to tweak. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi Sent: Monday, March 22, 2010 6:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] voicemail problem Hi people! I am running Asterisk 1.6.1.12 and I set up the voicemail. That far I have set upt the voicemailbox with my personal greeting message. If somebody calls me and is forwarded to my mailbox, my personal recorded greeting is played back + the default message please record your message after the tone and hang up or press the pound key. Is there a way to delete the second part from the voicemail, that only my personal recorded message is played back and a signal tone comes to signal the caller to start talking?! Tamer Higazi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP signal through one IP and media through different IPs
On 3/20/2010 10:34 AM, bruce bruce wrote: Hi Everyone, I have a provider who is asking me to send SIP signals through 111.111.111.111 and then media through Media 1: 222.222.22.222 and Media 2: 244.244.244.244. This provider authenticates by IP and I think is using Sonus gear and hence they have some load balancer or something... I have always simply done this to work it out: host=111.111.111.111 peer=type and everything worked. But now when I do that I have no audio with call established. I think it's a problem of me not assigning the media IPs. How can I add those to the trunk settings? Thanks, Bruce The media information is negotiated as part of the SIP call setup process, so I don't know of any way (or any reason) to force the media IPs. Some likely culprits would be incorrect firewall rules or something mangling the SDP information - I've seen something similar when the ip_nat_sip kernel module was loaded on a firewall between an Asterisk server and one of our SIP providers. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail problem
Hi Use the us option and not b _/List with the possible options/_ /*s* - _without_ this option a message will be played. The message by default is: Please leave your message after the tone. When done, hang up, or press the pound key. If you _set_ this option, the message won#8217;t be played. *u* - If you set this option, an unavailable message will be played. The message by default is: The person at extension dialled extension is unavailable. Also you will hear and the instructions: Please leave your message after the tone. When done, hang up, or press the pound key. *b* - If you set this option, a busy message will be played. The message by default is: The person at extension dialled extension is on the phone. Also you will hear and the instructions: Please leave your message after the tone. When done, hang up, or press the pound key. *su* - You will hear the unavailable message: The person at extension dialled extension is unavailable. The instruction message will be skipped. *sb* - You will hear the busy message: The person at extension dialled extension is on the phone. The instruction message will be skipped./ Ish Tamer Higazi wrote: Hi people! I am running Asterisk 1.6.1.12 and I set up the voicemail. That far I have set upt the voicemailbox with my personal greeting message. If somebody calls me and is forwarded to my mailbox, my personal recorded greeting is played back + the default message please record your message after the tone and hang up or press the pound key. Is there a way to delete the second part from the voicemail, that only my personal recorded message is played back and a signal tone comes to signal the caller to start talking?! Tamer Higazi -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get Asterisk to make batch calls?
On 3/21/2010 8:52 AM, Leo Burd wrote: Hello there, I'm currently building a PHP-based software to help users make batch calls. Basically, users provide a script and list of phone numbers. The system calls those numbers and plays the script to whoever picks up the phone. Currently, the system does one call at a time via direct access to the Asterisk Manager Interface, but does not seem to be very efficient. Ideally, I'd like to be able to run calls in parallel. What is the best way of doing that? Would I need to use some sort of proxy? Is this something that I should try to implement with AJAM? If possible, would anyone point me to code samples that might helpful to me? Thanks in advance, Leo Sounds like the simplest answer would be to use call files. http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call files : call multiple SIP-accounts
Hello, I'm trying to call different SIP-accounts to connect them to a conference. This is my call-file : Channel: SIP/test3SIP/test1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: from-conf Extension: 1000 I get the following in the CLI : [Mar 22 14:40:26] -- Attempting call on SIP/test3SIP/test1 for 1...@from-conf:1 (Retry 1) [Mar 22 14:40:26] WARNING[29908]: chan_sip.c:2994 create_addr: No such host: test3SIP [Mar 22 14:40:26] NOTICE[29908]: channel.c:3046 __ast_request_and_dial: Unable to request channel SIP/test3SIP/test1 [Mar 22 14:40:26] NOTICE[29908]: pbx_spool.c:356 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) So how can I simultaneously call different SIP-accounts from a call-file ?? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get Asterisk to make batch calls?
Call files are the simplest, but not the best answer. The best answer is to use the AMI in Asyncronous mode (Don't remember the exact syntax, but it is in the last 2-3 months of messages). The default behavior of AMI is one at a time calling, but this flag let's you push several calls through at once. Call files are a brute force solution, AMI is a more controlled one. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Sandy Sent: Monday, March 22, 2010 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Leo Burd Subject: Re: [asterisk-users] How to get Asterisk to make batch calls? On 3/21/2010 8:52 AM, Leo Burd wrote: Hello there, I'm currently building a PHP-based software to help users make batch calls. Basically, users provide a script and list of phone numbers. The system calls those numbers and plays the script to whoever picks up the phone. Currently, the system does one call at a time via direct access to the Asterisk Manager Interface, but does not seem to be very efficient. Ideally, I'd like to be able to run calls in parallel. What is the best way of doing that? Would I need to use some sort of proxy? Is this something that I should try to implement with AJAM? If possible, would anyone point me to code samples that might helpful to me? Thanks in advance, Leo Sounds like the simplest answer would be to use call files. http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP codec negotiation / manipulation
On 3/21/2010 4:05 AM, Olle E. Johansson wrote: 17 mar 2010 kl. 16.37 skrev Kevin Sandy: We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second invite requesting G729. However, they proceed to send us a G711 encoded audio stream... They have somewhat acknowledged the problem, but their advice is for us to only accept a single codec in our 200 OK. We don't want to disable either; we have customers using G729, so we'd like to avoid transcoding when possible, but we also do some T38 faxing, which I believe requires G711 to start off. My first thought was to selectively force the codec on inbound calls - if it is for a voice number, use 729, otherwise 711. However, I can't find any way of doing this within Asterisk. (We do have an OpenSIPS server sitting between us and the provider, and I could use OpenSIPS features to do this; however, right now the OpenSIPS server is fairly dumb - it's only proxying traffic between us and the provider and knows nothing about our specific DIDs.) A couple more details in case anyone has seen a similar issue. The provider is Broadvox, and this issue only seems to manifest on calls coming to them via Skype. They claim to not have any direct link with Skype, but it seems odd that the problem would be specific to Skype callers if the call is coming to Broadvox as a standard PSTN call. Is there any way to do this? Am I totally missing something and making a stupid mistake, or making the issue more complicated than it needs to be? The problem here is that you have a proxy in between, so Asterisk can't have separate peer configurations, since all the SIP messages are from the same IP and thus the same peer. I have a branch that implements peer matching in this specific configuration, which means that you can have different codec configurations for different partners even though there's a proxy in front of Asterisk. https://origsvn.digium.com/svn/asterisk/team/oej/pinetree-1.4 Please try this branch and give feedback. There should be some docs in sip.conf for the new matchrule setting. /O I'd be interested in trying this out - but the site doesn't seem to be responding. :) I have a few more questions, but I'm guessing that I can figure them out on my own once I have the code. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Context vs. Custom Context
Dear all, if I use the CustomContext module in Asterisk in order to create new customized contexts for my extensions to managed outbound/inbound calls, do these custom contexts replace the original context defined in sip.conf, like context=from-internal ??? In other words, does a custom context have a bigger priority than context ??? Thanks a lot, Alejandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call files : call multiple SIP-accounts
Not too long ago I needed to do the same thing but apparently you need to have a separate call file for every call. The dial command didn't work with an '' separating multiple destinations. I did it through a php script running via agi. On 2010-03-22 9:56 AM, jonas kellens jonas.kell...@telenet.be wrote: Hello, I'm trying to call different SIP-accounts to connect them to a conference. This is my call-file : Channel: SIP/test3SIP/test1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: from-conf Extension: 1000 I get the following in the CLI : [Mar 22 14:40:26] -- Attempting call on SIP/test3SIP/test1 for 1...@from-conf:1 (Retry 1) [Mar 22 14:40:26] WARNING[29908]: chan_sip.c:2994 create_addr: No such host: test3SIP [Mar 22 14:40:26] NOTICE[29908]: channel.c:3046 __ast_request_and_dial: Unable to request channel SIP/test3SIP/test1 [Mar 22 14:40:26] NOTICE[29908]: pbx_spool.c:356 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) So how can I simultaneously call different SIP-accounts from a call-file ?? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi Confusion
Dear community,
Please help. I've been looking around the internet (and in this great forum)
for help with DUNDi setup between servers (I'm using Elastix) and while I
can get my servers to lookup extensions on each other very well, I have not
been able to successfully make calls between servers. For my test
environment, I have 3 servers setup for now, and these are the steps I've
followed:
1. I edited dundi.conf on each server to have the following info:
(this listing is for all servers)
[mappings]
priv = ext-dundi,0,IAX2,priv:${SECRET}@
'server-hostname'/${NUMBER},nopartial
[00:1C:C0:65:34:04]
model = symmetric
host = 192.168.1.128
inkey = priv
outkey = priv
include = all
permit = all
qualify = yes
order = primary
dynamic = yes
[08:00:27:57:6E:0E]
model = symmetric
host = elastix-1
inkey = priv
outkey = priv
include = all
permit = all
qualify = yes
[08:00:27:15:0E:F1]
model = symmetric
host = elastix-2
inkey = priv
outkey = priv
include = all
permit = all
qualify = yes
order = primary
dynamic = yes
2. I also edited extensions_custom.conf in each server to have:
[ext-dundi]
include = ext-local
include = ext-paging
include = ext-intercom-users
include = ext-group
include = ext-meetme
3. I also created an IAX2 Trunk called 'priv' using FreePBX (placing
information below only within the PEER Details(this trunk shows up as
'IAX2/priv' in FreePBX/Elastix web configurator):
[priv]
type=friend
dbsecret=dundi/secret
context=from-internal
trunk=yes
4. I also created a DUNDi Trunk called 'priv' as well in FreePBX and edited
only the DUNDI Mapping in there. This too shows up as 'DUNDi/priv' in the
FreePBX/Elastix web configurator.
The next steps to do is what confuses me. My DUNDi lookups and queries work
fine, and I have no firewalls between the boxes.
I have created a route called dundi-outside in each server's FreePBX that
references the DUNDi/priv route, and subsequently deleted it, because
whenever i try to make calls i get either an 'all-circuits-are-busy' error
msg, or i get a
'call-cannot-be-completed-as-dialled-please-check-the-number-and-dial-again'
error.
I'm really confused as what is going wrong. Am I (surely) missing something?
Any help will be greatly appreciated.
Hope to hear from you soon.
--
best regards,
Sina Owolabi
2348034022578
23417203257
23417420690
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Re: [asterisk-users] Context vs. Custom Context
Alejandro Cabrera Obed wrote: Dear all, if I use the CustomContext module in Asterisk in order to create new customized contexts for my extensions to managed outbound/inbound calls, do these custom contexts replace the original context defined in sip.conf, like context=from-internal ??? In other words, does a custom context have a bigger priority than context ??? That sounds like a module for FreePBX or some other GUI. A context in Asterisk is just a context. There are no weights. If you define the same context twice you will likely get some sort of WARNING on the Asterisk console I think. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Context vs. Custom Context
Yes, Custom Context is a module from FreePBX in order to define calling routes. Thanks. 2010/3/22 Leif Madsen leif.mad...@asteriskdocs.org Alejandro Cabrera Obed wrote: Dear all, if I use the CustomContext module in Asterisk in order to create new customized contexts for my extensions to managed outbound/inbound calls, do these custom contexts replace the original context defined in sip.conf, like context=from-internal ??? In other words, does a custom context have a bigger priority than context ??? That sounds like a module for FreePBX or some other GUI. A context in Asterisk is just a context. There are no weights. If you define the same context twice you will likely get some sort of WARNING on the Asterisk console I think. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] too much sockets open by asterisk
Hello: 于 2010年03月20日 23:21, Leif Madsen 写道: CHEN XUEQIN wrote: I have a similar problem when using AGI for call control. Also udp port leak for some incomplete call. I wonder if the problem is related to issue 16774. Only way to know would be to reproduce on a development machine, and then try testing the patch on 16774 to see if the issue goes away. Patch 20100315__issue16774.diff.txt works. I found rtp port will disappear 32 seconds after the end of call. It's problem of dialog reference in code. Thanks. Regards, Xueqin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transcoding question
We are getting ready to install a client that uses g729 when talking to their SIP provider to minimize bandwidth usage. We are going to want to be able to record the calls using AMI monitor actions into wav sound files. All the phones are soft phone running on Windows XP systems. Questions I have are what would the best codec be to have the soft phone use since, as I understand it, in order to mix the audio something will need to be transcoded. Can a two CPU quad core xeon 2GHz system handle the transcoding load or would if be better to have a daughter card handle the transcoding. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] requirecalltoken receiving IAX calls
Hi All; I am configuring IAX endpoint, I just need to understand why I have to set requirecalltoken = no to be able to register because the following message is displayed: [Mar 22 12:25:39] ERROR[2297]: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 78.154.240.146 in the calltokenignore list or setting user gwbilalkwpciax requirecalltoken=no From the other side: if I make requirecalltoken=no , what does it mean? I am afraid if I make requirecalltoken=no then I will not be able to receive a calls on my IAX client, any advise? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] requirecalltoken receiving IAX calls
There was some security patch that changed some details of the IAX protocol. There are now some tokens used to make the connection more secure or some such thing. If you have some newer and older versions of asterisk that want to talk to each other you need to tell the newer versions to not expect the older versions to know about tokens. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Mar 22, 2010, at 10:26 AM, bilal ghayyad wrote: Hi All; I am configuring IAX endpoint, I just need to understand why I have to set requirecalltoken = no to be able to register because the following message is displayed: [Mar 22 12:25:39] ERROR[2297]: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 78.154.240.146 in the calltokenignore list or setting user gwbilalkwpciax requirecalltoken=no From the other side: if I make requirecalltoken=no , what does it mean? I am afraid if I make requirecalltoken=no then I will not be able to receive a calls on my IAX client, any advise? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] requirecalltoken receiving IAX calls
On 22 Mar 2010, at 17:26, bilal ghayyad wrote: From the other side: if I make requirecalltoken=no , what does it mean? I am afraid if I make requirecalltoken=no then I will not be able to receive a calls on my IAX client, any advise? http://lmgtfy.com/?q=asterisk+config+iax.conf+requirecalltoken Top link explains it. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Context vs. Custom Context
Alejandro Cabrera Obed wrote: Yes, Custom Context is a module from FreePBX in order to define calling routes. I'd suggest using the FreePBX forums as I imagine the majority of people responding on this list are vanilla Asterisk users. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail, Asterisk and Grandstream BT200
you are right, under [channels] is where it's supposed to be my mistake, i guess i was thinking in sip.conf :) However, the following doubt arises to me: it would also have had this problem for some originating call from a telephone that is not a cell phone? yes, and this can be a really serious problem if you don't fix it. So don't forget to include this parameters from now on. I have played with them and found setting busycount=5 is not very efficent, so leave it to 3 or 4 at most. Good to hear your problem is solved. Alyed 2010/3/22 Daniel Bareiro daniel-lis...@gmx.net -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Alyed. On Mon, 22 Mar 2010, Alyed wrote: I was with the following situation: if I call from a cell phone, my Asterisk take the call, it presents to the caller the possibility to dialing an extension number and, in case of not doing it, it transfers this call to a specific extension. Then, if in this extension nobody takes the call, the service of voicemail is triggered so that the caller leaves its message from the cell phone. But if it hangs after to let the message without have pressed previously the pound key, the channel is taken and no longer any other call enters the PBX from the PSTN. This does not happen if the caller presses the pound key after to have left his message. As I have a box at which the cable arrives from the PSTN in which there are two ports of derivation and in one of them it leaves the cable for the Asterisk PBX (connected only then), after to have detected this problem I tried connecting in the other port an analog telephone and, indeed, it did not have tone as if never it had been hung. In addition this was confirmed because the MWI light never blinked on the telephone. After restarting the Asterisk server, yes the MWI light blinks and in addition I could corob the time in which the channel was taken seeing that the message lasted more than nine minutes. To what this problem can be due? It has to do the call is made specifically from cell phone through the PSTN (because if I leave a message hanging directly without pressing the pound key from an local extension, this does not happen)? There is some form to avoid it? Make sure you have busydetect=yes busycount=3 somewhere below your [general] context in chan_dahdi.conf (or zapata.conf depending on your asterisk version) and restart the the service. This should be enoough to do the magic. It didn't have configured these two parameters so I added now them but in the [channels] context since I don't have a [general] context (It does not sound to me that in the file by default generated by Asterisk there would not be it either, although I can be mistaken). Beyond that, with these two parameters, I no longer have the problem mentioned before. Thanks! However, the following doubt arises to me: it would also have had this problem for some originating call from a telephone that is not a cell phone? Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkunNjQACgkQZpa/GxTmHTfAbACfT8PVkcp/xESdqsiczg3YY/Dd FGcAn1TdOqiZaKAjLg4h3SDt/34A4bKX =37qZ -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Play music to caller after answer, before dial
I would like to play music to an inbound caller, AFTER asterisk answers the call, but before the call is bridged by DIAL. Is there a simple way to achieve this? MD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free iPhone Asterisk Function and Application Reference
On 20/03/10 9:47 AM, Tzafrir Cohen wrote: On Fri, Mar 19, 2010 at 10:50:17AM -0400, Zeeshan Zakaria wrote: Hi Matt, This is very useful. But what about android platforms? Will it run on it? Just use an RSS reader. I guess browsers and RSS readers on the iPhone are too limited. There are a couple, the main diff with the Daily Asterisk News app is that you can vote, see similar articles etc. But no, it doesn't run on Android and I have no plans to write it for Android just yet :) Still getting to know iPhone/iPad SDK - once done I might start learning Android SDK :) -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free Daily Asterisk News iPhone and iPod Touch app
On 20/03/10 3:46 AM, Zeeshan Zakaria wrote: Actually I might be wrong but haven't tried it yet because the download page is not available or the link is broken. I have however an iPhone too to try it. Which link is broken? I just clicked on it from the original email, and then clicked the download link. When done from an iPhone it brings up the app with a link to download, on my Mac it opens iTunes to the application page. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Play music to caller after answer, before dial
This might be your answer: Exten = s,1,answer Exten = s,n,wait(10,m) Exten = s,n,Dial. This would wait 10 seconds playing MOH before dialing. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Monday, March 22, 2010 3:58 PM To: 'Asterisk Users List' Subject: [asterisk-users] Play music to caller after answer, before dial I would like to play music to an inbound caller, AFTER asterisk answers the call, but before the call is bridged by DIAL. Is there a simple way to achieve this? MD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk running on a Xen Centos Serverchallenge!!!
There's something weird at you linux system. Your kernel sources are i686 sources, but the output of your ls shows you are on a X86-64 system. (?) lrwxrwxrwx 1 root root 54 Nov 6 23:31 build - ../../../usr/src/kernels/2.6.18-164.6.1.el5-xen-x86_64 Look here! -bash-3.2# ls /usr/src/kernels 2.6.18-164.6.1.el5-xen-i686 2.6.18-164.6.1.el5xen-i686 and here! To install (and update) the correct kernel and kernel sources, try using: yum install kernel-* After that, reboot with the new kernel and recompile Dahdi. Rafael Prado +55 (11) 3323-1055 ttp://www.practis.com.br PRACTIS - Comunicação Tecnologia Av Aquidaban, 766 - Conj 51 CEP 13026-510, Campinas/SP - Brasil -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Daniel Leite de Abreu Sent: domingo, 21 de março de 2010 10:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk running on a Xen Centos Serverchallenge!!! Hi Thanks very much for reply it and helping me out. This is the out put -bash-3.2# ls -l /lib/modules/2.6.18-164.6.1.el5xen/ total 1280 lrwxrwxrwx 1 root root 54 Nov 6 23:31 build - ../../../usr/src/kernels/2.6.18-164.6.1.el5-xen-x86_64 drwxr-xr-x 2 root root 4096 Nov 3 17:31 extra drwxr-xr-x 9 root root 4096 Nov 6 23:31 kernel -rw-r--r-- 1 root root 282675 Nov 6 23:31 modules.alias -rw-r--r-- 1 root root 69 Nov 6 23:31 modules.ccwmap -rw-r--r-- 1 root root 231095 Nov 6 23:31 modules.dep -rw-r--r-- 1 root root147 Nov 6 23:31 modules.ieee1394map -rw-r--r-- 1 root root375 Nov 6 23:31 modules.inputmap -rw-r--r-- 1 root root 12632 Nov 6 23:31 modules.isapnpmap -rw-r--r-- 1 root root 74 Nov 6 23:31 modules.ofmap -rw-r--r-- 1 root root 219500 Nov 6 23:31 modules.pcimap -rw-r--r-- 1 root root 4033 Nov 6 23:31 modules.seriomap -rw-r--r-- 1 root root 132264 Nov 6 23:31 modules.symbols -rw-r--r-- 1 root root 356940 Nov 6 23:31 modules.usbmap lrwxrwxrwx 1 root root 5 Nov 6 23:31 source - build drwxr-xr-x 2 root root 4096 Nov 3 17:31 updates drwxr-xr-x 2 root root 4096 Nov 3 17:31 weak-updates -bash-3.2# This is the other out put. -bash-3.2# ls /usr/src/kernels 2.6.18-164.6.1.el5-xen-i686 2.6.18-164.6.1.el5xen-i686 waiting for you . Thanks very much daniel On 19 Mar 2010, at 8:51 PM, Tzafrir Cohen wrote: On Fri, Mar 19, 2010 at 01:26:43AM +0200, Tzafrir Cohen wrote: On Thu, Mar 18, 2010 at 05:03:12PM -0500, Warren Selby wrote: On Thu, Mar 18, 2010 at 6:56 PM, Daniel Leite de Abreu dlab...@gmail.comwrote: Hi David! Thanks very much for helping me out will all ! Ok i try your tip and @ the moment i still have the same problem but now i have the kernel and the kernel devel the same but wend i try to run make i still get the same erro, do you guys have any idea how to fix it? -bash-3.2# rpm -qa kernel* kernel-xen-devel-2.6.18-164.6.1.el5 kernel-xen-2.6.18-164.6.1.el5 -bash-3.2# After you install the kernel source, you'll need to rerun ./configure. Nope. The dahdi-linux makefile has no ./configure . You may want to run make clean and / or make distclean before rerunning ./configure. Specifically: it will look for: /lib/modules/VERSION/build/.config Where: VERSION is the kernel version string. 2.6.18-164.6.1.el5 in your case. 'build' is a symbolic link to the (often partial) kernel tree. What is the output of: ls -l /lib/modules/2.6.18-164.6.1.el5 What is the output of: ls /usr/src/kernels -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning
I know for a fact that you can provision a Polycom via ftp. I've included much of my dhcpd.conf file below. Pick out what you need. Let me know if I can confirm that using option 66 will work with FTP (and HTTP, for that matter) with newer BootROM versions. I don't know the exact version it changed, unfortunately, as I just noticed it in passing when I was running some tests one day. As for why we ended up choosing option 129 originally rather than 160? I wish I had a clever technical explation, but it's just a random unassigned option number. That's it. :) - Brad Thanks. This has been very helpful. It appears that without specifying the protocol in the option 66 host string, Polycom endpoints default to TFTP even if you specify FTP in the Polycom bootloader. This appears to be the case EVEN IF you tell the bootloader to expect an IP address (instead of a string which can naturally stipulate the protocol). For us (and apparently others), this idiosyncracy turned into a problem in that OUR DHCP server (pfSense 1.2.3--a gui around dhcpd) can only pass an IP address, not a host string (with protocol, credentials, etc). in DHCP option 66. A workaround for this could be achieved (in pfSense 1.2.3) by uploading a special configuration file for dhcpd, (For option 66 or ANY other DHCP option for that matter) but this special config can be easily overwritten by the pfSense gui, so it should be considered maintenece gotcha/problem ala Oops, nobody's phone works after this unrelated change It appears that pfSense 2.0 (currently in beta) will offer custom DHCP options within the GUI. FWIW, we have decided to pre-configure new endpoints with the local DNS name of the provisioning server, as well as the provisioning protocol, username (the default) and secret. This means that phones would not need to be revisited in order to point the endpoints to another provisioning server. -Karl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Play music to caller after answer, before dial
I think I forgot some important information... I'm actually running an AGI script after the answer (and before the dial). I would like to play MOH while the AGI script is running, and then perform the dial (ending the MOH). This is where I'm stuck Thanks! Michelle _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, March 22, 2010 5:22 PM To: Asterisk Users List Subject: Re: [asterisk-users] Play music to caller after answer, before dial This might be your answer: Exten = s,1,answer Exten = s,n,wait(10,m) Exten = s,n,Dial. This would wait 10 seconds playing MOH before dialing. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Monday, March 22, 2010 3:58 PM To: 'Asterisk Users List' Subject: [asterisk-users] Play music to caller after answer, before dial I would like to play music to an inbound caller, AFTER asterisk answers the call, but before the call is bridged by DIAL. Is there a simple way to achieve this? MD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using asterisk as avaya definity recordingserver
Hi, it's not that simple.
It requires deep modification on asterisk and dahdi sources to work the way
you want.
Rafael Prado
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Muro, Sam
Sent: quarta-feira, 17 de março de 2010 1:42
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Using asterisk as avaya definity
recordingserver
Hi there
Looks like someone hasnt done this!! I have been thinking and find out
that Monitor/Spy and the likes wont help me as the call need to be
bridged with the asterisk core or via channel drivers.
My final shot now is on Record() function. Since the legacy system will
forward the call to the monitoring interfaces when bridged within
itself, it the interface in on Asterisk, then we can capture the
pattern and use
exten =
#CALLER_NUMBER#CALLED_NUMBER,1,Record(/var/spool/asterisk/monitor/avaya
-${EXTEN:1:4}-${EXTEN:4:4}:wav)
This assume that Len(CALLER_NUMBER) = 4
Anyone with alternative solution?
Muro, Sam wrote:
Oh.. I didnt know that.
Thanks
Sam
Muro, Sam escribió:
What do you mean chief? What am looking at is ability for asterisk
to
receive a call and recording until it tier down without bridging it
to the physical device
Sam
Would you like the advice in all caps?
He means that you put the subject in all caps. He normally gets
upset
with everyone that does this on the subject or in the body. I've
corrected the caps in the subject to avoid further upsetting.
Cheers,
--
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Description: S/MIME cryptographic signature
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Re: [asterisk-users] Transcoding question
How many simultaneous channels? Rafael Prado -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jim Dickenson Sent: segunda-feira, 22 de março de 2010 2:33 To: Asterisk User MailList Subject: [asterisk-users] Transcoding question We are getting ready to install a client that uses g729 when talking to their SIP provider to minimize bandwidth usage. We are going to want to be able to record the calls using AMI monitor actions into wav sound files. All the phones are soft phone running on Windows XP systems. Questions I have are what would the best codec be to have the soft phone use since, as I understand it, in order to mix the audio something will need to be transcoded. Can a two CPU quad core xeon 2GHz system handle the transcoding load or would if be better to have a daughter card handle the transcoding. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PGSQL application
Or you can write your own application module. Try looking at cdr_pgsql sources. ;) Rafael Prado +55 (11) 3323-1055 www.practis.com.br PRACTIS - Comunicação Tecnologia Av Aquidaban, 766 - Conj 51 CEP 13026-510, Campinas/SP - Brasil -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Vinícius Fontes Sent: quarta-feira, 10 de março de 2010 8:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PGSQL application - Tilghman Lesher tles...@digium.com escreveu: On Wednesday 10 March 2010 14:32:56 Vinícius Fontes wrote: Does the application PGSQL has been removed from Asterisk? Couldn't find it on Asterisk source and addons. That application has never been a part of Asterisk in the first place. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org Hmm I could swear I used it on the 1.2 days. So, in order to access PostgreSQL directly from the dialplan without the use of AGIs, much like the MYSQL() app, the only way to go is via the function ODBC()? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can I call myself on the same machine
I am a newbie to asterisk, I have a complete installation of asterisk running on my ubuntu machine and I have x-lite installed also, I would like to know if I can call myself on the same machine, because whenever I try to call myself I get an engaged tone. My configuration file settigns are as below: sip.conf: [general] context=default srvlookup=yes [1000] type=friend secret=welcome qualify=yes nat=yes host=dynamic canreinvite=yes context=internal extensions.conf: [globals] [general] autofallthrough=yes [default] exten = s,1,Verbose(1|Unrouted call handler) exten = s,n,Answer() exten = s,n,Wait(1) exten = s,n,Playback(tt-weasels) exten = s,n,Hangup() [incoming_calls] [internal] exten = 100,1,Dial(SIP/john) exten = 611,1,Echo( ) [phones] include = internal and when I run sip show peers on my CLI console I get: Name/username HostDyn Nat ACL Port Status 1000 (Unspecified)D N 5060 UNKNOWN 1 sip peers [Monitored: 0 online, 1 offline Unmonitored: 0 online, 0 offline] I would appreciate soonest response to the mail. Best Regards _ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding question
There will be up to 150 phones so there will be 300 channels when they are all on the phone at one time. I will be using a current 1.4 version. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Mar 22, 2010, at 5:05 PM, Rafael Prado Rocchi wrote: How many simultaneous channels? Rafael Prado -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jim Dickenson Sent: segunda-feira, 22 de março de 2010 2:33 To: Asterisk User MailList Subject: [asterisk-users] Transcoding question We are getting ready to install a client that uses g729 when talking to their SIP provider to minimize bandwidth usage. We are going to want to be able to record the calls using AMI monitor actions into wav sound files. All the phones are soft phone running on Windows XP systems. Questions I have are what would the best codec be to have the soft phone use since, as I understand it, in order to mix the audio something will need to be transcoded. Can a two CPU quad core xeon 2GHz system handle the transcoding load or would if be better to have a daughter card handle the transcoding. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can I call myself on the same machine
On Mon, 2010-03-22 at 23:40 +, ayodele abejide wrote: I am a newbie to asterisk, I have a complete installation of asterisk running on my ubuntu machine and I have x-lite installed also, I would like to know if I can call myself on the same machine, because whenever I try to call myself I get an engaged tone. My configuration file settigns are as below: If you plan to use a softphone on the same machine then you need to tell it to use a different port to listen for SIP. Asterisk and your softphone are both trying to listen to port 5060, that is why it always fails. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can I call myself on the same machine
I tried port 5061 for my softphone, but the same problem occurs From: cur...@telecomabmex.com To: asterisk-users@lists.digium.com Date: Mon, 22 Mar 2010 17:54:27 -0600 Subject: Re: [asterisk-users] Can I call myself on the same machine On Mon, 2010-03-22 at 23:40 +, ayodele abejide wrote: I am a newbie to asterisk, I have a complete installation of asterisk running on my ubuntu machine and I have x-lite installed also, I would like to know if I can call myself on the same machine, because whenever I try to call myself I get an engaged tone. My configuration file settigns are as below: If you plan to use a softphone on the same machine then you need to tell it to use a different port to listen for SIP. Asterisk and your softphone are both trying to listen to port 5060, that is why it always fails. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Play music to caller after answer, before dial
Does this help? The A near the end calls the audio file ginr3 exten = 551,1,Answer() exten = 551,n,Dial(SIP/callwithus/17025551212,120,A(ginr3)) On Mon, Mar 22, 2010 at 6:41 PM, Michelle Dupuis mdup...@ocg.ca wrote: I think I forgot some important information... I'm actually running an AGI script after the answer (and before the dial). I would like to play MOH while the AGI script is running, and then perform the dial (ending the MOH). This is where I'm stuck Thanks! Michelle From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, March 22, 2010 5:22 PM To: Asterisk Users List Subject: Re: [asterisk-users] Play music to caller after answer, before dial This might be your answer: Exten = s,1,answer Exten = s,n,wait(10,m) Exten = s,n,Dial… This would wait 10 seconds playing MOH before dialing. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Monday, March 22, 2010 3:58 PM To: 'Asterisk Users List' Subject: [asterisk-users] Play music to caller after answer, before dial I would like to play music to an inbound caller, AFTER asterisk answers the call, but before the call is bridged by DIAL. Is there a simple way to achieve this? MD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which folder for sounds?
1.6.2: -- Executing [...@incoming-pstn-line:4] VoiceMail(DAHDI/4-1, 1...@default,u) in new stack -- DAHDI/4-1 Playing '/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en') [Mar 22 17:15:46] WARNING[31145]: file.c:650 ast_openstream_full: File vm-intro does not exist in any format [Mar 22 17:15:46] WARNING[31145]: file.c:953 ast_streamfile: Unable to open vm-intro (format 0x4 (ulaw)): No such file or directory But: locate vm-intro /var/lib/asterisk/sounds/en/vm-intro.gsm /var/lib/asterisk/sounds/en/vm-intro.ulaw /var/lib/asterisk/sounds/en/vm-intro.wav head -12 /etc/asterisk/asterisk.conf [directories](!) ; remove the (!) to enable this astetcdir = /etc/asterisk astmoddir = /usr/lib64/asterisk/modules astvarlibdir = /var/lib/asterisk astdbdir = /var/lib/asterisk astkeydir = /var/lib/asterisk astdatadir = /var/lib/asterisk astagidir = /var/lib/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk So in which folder are these sounds supposed to be? But more importantly, who do we figure out where * is looking for sounds? Some cli command would be really nice. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Play music to caller after answer, before dial
No...I need the audio to play while the AGI is running, but BEFORE the dial command. I did solve it eventually by using the Dial Local Channel...while the AGI is running and before the second answer Thanks MD -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron Sent: Monday, March 22, 2010 9:18 PM To: Asterisk Users List Subject: Re: [asterisk-users] Play music to caller after answer, before dial Does this help? The A near the end calls the audio file ginr3 exten = 551,1,Answer() exten = 551,n,Dial(SIP/callwithus/17025551212,120,A(ginr3)) On Mon, Mar 22, 2010 at 6:41 PM, Michelle Dupuis mdup...@ocg.ca wrote: I think I forgot some important information... I'm actually running an AGI script after the answer (and before the dial). I would like to play MOH while the AGI script is running, and then perform the dial (ending the MOH). This is where I'm stuck Thanks! Michelle From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, March 22, 2010 5:22 PM To: Asterisk Users List Subject: Re: [asterisk-users] Play music to caller after answer, before dial This might be your answer: Exten = s,1,answer Exten = s,n,wait(10,m) Exten = s,n,Dial This would wait 10 seconds playing MOH before dialing. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Monday, March 22, 2010 3:58 PM To: 'Asterisk Users List' Subject: [asterisk-users] Play music to caller after answer, before dial I would like to play music to an inbound caller, AFTER asterisk answers the call, but before the call is bridged by DIAL. Is there a simple way to achieve this? MD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which folder for sounds?
/var/lib/asterisk/sounds/ On 2010-03-22 9:44 PM, sean darcy seandar...@gmail.com wrote: 1.6.2: -- Executing [...@incoming-pstn-line:4] VoiceMail(DAHDI/4-1, 1...@default,u) in new stack -- DAHDI/4-1 Playing '/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en') [Mar 22 17:15:46] WARNING[31145]: file.c:650 ast_openstream_full: File vm-intro does not exist in any format [Mar 22 17:15:46] WARNING[31145]: file.c:953 ast_streamfile: Unable to open vm-intro (format 0x4 (ulaw)): No such file or directory But: locate vm-intro /var/lib/asterisk/sounds/en/vm-intro.gsm /var/lib/asterisk/sounds/en/vm-intro.ulaw /var/lib/asterisk/sounds/en/vm-intro.wav head -12 /etc/asterisk/asterisk.conf [directories](!) ; remove the (!) to enable this astetcdir = /etc/asterisk astmoddir = /usr/lib64/asterisk/modules astvarlibdir = /var/lib/asterisk astdbdir = /var/lib/asterisk astkeydir = /var/lib/asterisk astdatadir = /var/lib/asterisk astagidir = /var/lib/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk So in which folder are these sounds supposed to be? But more importantly, who do we figure out where * is looking for sounds? Some cli command would be really nice. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to make upgrades with Asterisk
Hello my friends, I want to make upgrades for all my software, currently i have the following versions: Asterisk 1.4.21.2 Zaptel Version: 1.4.11 WANPIPE Release: 3.4.7 libpri version: 1.4.5 I want to make upgrade for the last version of Asterisk 1.4, the last version of Zaptel (dahdi will be nice!), the newest libpri version and wanpipe What should i do? this is a production server and i don't want to mess something...is there a step by step to make this upgrade? Thanks in advance for your valuable help! DD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to make upgrades with Asterisk
If it is a production server, you should not do the upgrade on it. Setup a new server with upgraded software, migrate all the data, test it and make sure it works fine. There are things like CDR and voicemail which are constantly being updated, meaning just before the final migration, you should copy them to the new server. I have done some migrations and have found clonezilla to be a wonderful tool for this purpose. You create a second server on any computer, and once it is ready, clone it on a USB stick or CD, or on another computer via SSH, then clone the production server for backup purposes on a medium of your choice, and finally restore this new server image on to the production server. If something goes wrong, you'll be able to restore the server back to its functional state from the backup cloned image. When I migrated my own production server from 1.2 to 1.4, I did the rehearsal many times, and very carefully drafted the whole migation plan. This also included asking all the users to copy and delete their voicemails before the day of migration. It took me about two weeks in planning and making sure every single setting will be migrated, before I was comfotable to do the migration, which took hardly an hour, and went just perfectly smooth. Personally I am of the opinion that if it is not really necessary, don't upgrade it. Will 1.6 give you something which you don't have in 1.4? It'll have its own issues and learning curve. I tried it once and it was only a pain for my setup, specially with real-time architecture, and a few other things which I can't remember now. -- Zeeshan A Zakaria On 2010-03-22 10:48 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello my friends, I want to make upgrades for all my software, currently i have the following versions: Asterisk 1.4.21.2 Zaptel Version: 1.4.11 WANPIPE Release: 3.4.7 libpri version: 1.4.5 I want to make upgrade for the last version of Asterisk 1.4, the last version of Zaptel (dahdi will be nice!), the newest libpri version and wanpipe What should i do? this is a production server and i don't want to mess something...is there a step by step to make this upgrade? Thanks in advance for your valuable help! DD -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
-- 祝您愉快！！ Aaron Chen 陈江涛 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which folder for sounds?
Hi! [directories](!) ; remove the (!) to enable this You did see the above, did you. '/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en') [Mar 22 17:15:46] WARNING[31145]: file.c:650 ast_openstream_full: File vm-intro does not exist in any format Did you maybe upgrade from 1.2 to 1.6? Taken from UPGRADE.txt on a 1.4 system: WARNING: Asterisk 1.4 supports a new layout for sound files in multiple languages; instead of the alternate-language files being stored in subdirectories underneath the existing files (for French, that would be digits/fr, letters/fr, phonetic/fr, etc.) the new layout creates one directory under /var/lib/asterisk/sounds for the language itself, then places all the sound files for that language under that directory and its subdirectories. This is the layout that will be created if you select non- English languages to be installed via menuselect, HOWEVER Asterisk does not default to this layout and will not find the files in the places it expects them to be. If you wish to use this layout, make sure you put 'languageprefix=yes' in your /etc/asterisk/asterisk.conf file, so that Asterisk will know how the files were installed. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
