Re: [asterisk-users] Connect Asterisk to a cell phone
The cellphone can be presented to Asterisk as SIP device using OpenBTS (GSM to SIP conversion). On Tue, Feb 15, 2011 at 10:40 PM, Faisal Hanif fai...@vopium.com wrote: Hi, Your question is not clear but below are possible answers to your question, If you want to attach you cell-phone to asterisk you can simply use chan_mobile. Using Bluetooth with chan_mobile you can connect your Cell-Phone as FXO and your handsfree as FXS port to asterisk. If you are asking about a GSM to SIP gateway then yes there are number of product available that can hold 1-256 SIM and register as SIP gateway to asterisk for incoming and outgoing calls. If you are asking about GSM PCI card then also yes there are PCI cards available for GSM/CDMA/HSPDA for 1-16 SIMs. Can pluged to asterisk PBX machine and used as FXO device. Regards, Faisal Hanif *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *logan *Sent:* Wednesday, February 16, 2011 10:49 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Connect Asterisk to a cell phone Hello, Are there any gateways which allow me to hook a cellphone to Asterisk and use that line for routing my calls? Basically, I'm looking to play around a bit and if I can get to connect a cellphone with Asterisk then that would be great. Thanks, Hitesh PS: I have tried to search on the web, but didn't find any pointers on how to do so. -- _ -- 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] uptime
On Tue, 15 Feb 2011, Hans Witvliet wrote: On Tue, 2011-02-15 at 09:01 +, Steve Howes wrote: On 15 Feb 2011, at 03:39, Jeff LaCoursiere wrote: minipbx*CLI show uptime System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds Last reload: 8 hours, 3 minutes, 51 seconds What's the highest current 'genuine' one on-list?.. klein*CLI core show uptime System uptime: 2 years, 1 week, 4 days, 21 hours, 52 minutes Last reload: 41 weeks, 6 days, 16 hours, 6 minutes, 39 seconds That's the best I can come up with.. Got a good old sparc system (RedHat 6.4 for sparc) doing an imap server. Had a power dip, just before i got to 10 years up. Must be somewhere on the sparc mailing list archives. After that, i had to power-cycle it again to put it on the ups. I used to wory about uptimes, etc. these days I'm not so concerend, however: $ uptime 07:53:13 up 1231 days, 13:13, 2 users, load average: 0.00, 0.00, 0.00 is my longest running box - it's a router and running my own embedded linux variant, and been facing the internet for all that time, routing GB of data daily... No-doubt I'll now get a barrage of whinges from people telling me it's insecure because it's not been patched, etc. which seems to be the popular reason for people rebooting Linux servers, or just the MS mentality, however I've no plans to reboot it until it's retired - probably at the end of next year. It's one of a pair running HA and VRRP - and it's counterpart is a bit younger: $ uptime 07:54:14 up 841 days, 10:59, 1 user, load average: 0.00, 0.00, 0.00 I don't quite have the same uptime with my asterisk boxes - dsx$ uptime 08:00:07 up 275 days, 16:09, 1 user, load average: 0.00, 0.00, 0.00 that's in a busy little design office with 10 people. Gordon -- _ -- 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] Regarding error in asterisk 1.6.2.16....
Hi every one, When i run asterisk i am getting error as ERROR[2109]: chan_sip.c:3963 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data i am unable to understand what it is and how come.. can any one help me regarding this issue.. Hope i get a positive reply.. With Regards, viswavardhan -- _ -- 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] uptime
On Mon, Feb 14, 2011 at 10:39:20PM -0500, Jeff LaCoursiere wrote: Now this is what I call uptime... minipbx*CLI show uptime System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds Last reload: 8 hours, 3 minutes, 51 seconds Bizarre bug? Hi. I see that 41 years, 7 weeks,...,46 seconds is $ TZ=EST5EDT date -d @`dc -e '41 365*7 7*+6+24*3+60*26+60*46+p'` Mon Feb 14 22:26:46 EST 2011 about 13 min. before your post, meaning Asterisk apparently used 0 as its start time when calculating uptime. How do you set the system time? Is it possible that Asterisk starts before the time is set? Do you run ntpdate -b at startup, or set the time by some other means before the boot completes? Could this result in some really humorous CDRs? -- Barry -- _ -- 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 know Caller's last position in Queue?
Hi group,
I have a simple call center scenario set up on Asterisk. Customer calls the
DID and gets placed in Queue waiting for their turn to talk to the available
agent.
Sometimes Customer hangs up in between and in this case I want to get the
last position of customer in Queue.
I know there is a variable called ${QEORIGINALPOS} that gives us original
position of caller in Queue, but there doesn't seem to have something
similar for exit position.
Am I missing something?
Thanks,
--AsteriskMan
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[asterisk-users] Barge in.
I'm running Asterisk 1.6 and was wondering if anybody have a workig barge in solution running. I was thinking of using chanspy, but i would like that the original call would be dropped, and the new call would be the only one there. -- _ -- 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] Aastra phones cannot transfer calls?
On 16 February 2011 00:22, Ernie Dunbar maill...@lightspeed.ca wrote: At 12:12 PM 2/15/2011, you wrote: I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk v1.6.2.1. They can call each other's extensions (and make and receive calls otherwise), but they cannot transfer calls, not even to outside I'm running 1.6.2.16.1 and have three Aastra 480i phones and have had no problem at all with transfers. Have you considered trying a newer version? Nope. Upgraded to 1.6.2.16.1, and I still see the same effect. It may be a setting on the phone or a SIP setting. I'll investigate this elsewhere but report back about the solution. I also tried this with a 6757i and a 6753i with no problems (blind and attended) on Asterisk 1.6.2.16.1. Have you updated the handset firmware to 2.6.0.2010? Cheers, Steve -- _ -- 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] Barge in.
On 16 February 2011 10:13, Peter den Hartog peterdenhar...@gmail.com wrote: I'm running Asterisk 1.6 and was wondering if anybody have a workig barge in solution running. I was thinking of using chanspy, but i would like that the original call would be dropped, and the new call would be the only one there. What you are describing looks to me like a third party controlled transfer, and not a barge-in at all. I suspect that the Asterisk Manager API action Redirect will be your friend. Regards, Steve -- _ -- 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] Cisco 7945G phone with asterisk
Hi, Anyone who has deployed Cisco 7945G phone with asterisk, kindly share your experience. /ag -- _ -- 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] Barge in.
Okay, so let me try to make it more clear to be sure everybody gets it :-), i can be a bit unclear from time to time ;-). 100 is in a call with 101. 102 has a higher priority and calls 100. The call between 100 101 disconnects, and 102 100 are connected. Peter On Wed, Feb 16, 2011 at 11:29 AM, Steve Davies davies...@gmail.com wrote: On 16 February 2011 10:13, Peter den Hartog peterdenhar...@gmail.com wrote: I'm running Asterisk 1.6 and was wondering if anybody have a workig barge in solution running. I was thinking of using chanspy, but i would like that the original call would be dropped, and the new call would be the only one there. What you are describing looks to me like a third party controlled transfer, and not a barge-in at all. I suspect that the Asterisk Manager API action Redirect will be your friend. Regards, Steve -- _ -- 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] Detect #,* DTMF in dialplan
Dear Mr,Ms; I am planing for a custom IVR, for example to act as a simple installer! I mean there is some choice via 0-9 and # as *Next* and * as *Back* button. is there any way for me to detect if the caller pressed # vs * on Dialplan ? -- Regards, Ali R. Taleghani 0936 322 4069 -- _ -- 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] Detect #,* DTMF in dialplan
Try it with your own AGI-Script - this is more flexible. http://www.voip-info.org/wiki/view/Asterisk+AGI Am 16.02.2011 11:45, schrieb shayne.al...@gmail.com: Dear Mr,Ms; I am planing for a custom IVR, for example to act as a simple installer! I mean there is somechoicevia 0-9 and # as Next and * as Backbutton. is there any way for me to detect if the caller pressed # vs * on Dialplan ? -- Regards, Ali R. Taleghani 0936 322 4069 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to know Caller's last position in Queue?
If you use Asterisk 1.8.x you can have this in channel vars and can collect
and add to DB or file on h extension.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Man
Sent: Wednesday, February 16, 2011 3:06 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to know Caller's last position in Queue?
Hi group,
I have a simple call center scenario set up on Asterisk. Customer calls the
DID and gets placed in Queue waiting for their turn to talk to the available
agent.
Sometimes Customer hangs up in between and in this case I want to get the
last position of customer in Queue.
I know there is a variable called ${QEORIGINALPOS} that gives us original
position of caller in Queue, but there doesn't seem to have something
similar for exit position.
Am I missing something?
Thanks,
--AsteriskMan
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Cisco 7945G phone with asterisk
On Wed, Feb 16, 2011 at 7:32 AM, ast guy ast...@gmail.com wrote: Hi, Anyone who has deployed Cisco 7945G phone with asterisk, kindly share your experience. /ag I did some 7941's a few months ago with SIP. They work pretty well. Make a console cable for the AUX port and you can see them load. I had to add Spanish menus to them so I ended up hacking the load process (Cisco does not support language files with SIP firmware). The 7945 should just have more features. ~~~ Andrew lathama Latham lath...@gmail.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] how to diable echo cancellation for sip?
Hello, can anyboby tell me, how can I disable the echo cancellation for sip? thx a lot... best regards, Felix -- _ -- 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] Connect Asterisk to a cell phone
You might use a SIP-to-cellular gateway as I on did. http://www.mgraves.org/2008/11/how-to-add-a-cellular-trunk-to-your-voip- system-part-1/ Michael --Original Message Text--- From: logan Date: Tue, 15 Feb 2011 21:49:26 -0800 Hello, Are there any gateways which allow me to hook a cellphone to Asterisk and use that line for routing my calls? Basically, I'm looking to play around a bit and if I can get to connect a cellphone with Asterisk then that would be great. Thanks, Hitesh PS: I have tried to search on the web, but didn't find any pointers on how to do so. -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- 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] function Echo() doesn't work
Hi guys, the function Echo() did work on CAPI, but doesn't work for SIP connection. Can anybody help? thanks a lot. best regards, Felix -- _ -- 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] Connect Asterisk to a cell phone
--Original Message Text--- From: Michael Graves Date: Wed, 16 Feb 2011 05:46:30 -0600 You might use a SIP-to-cellular gateway as I on did. http://www.mgraves.org/2008/11/how-to-add-a-cellular-trunk-to-your-voip- system-part-1/ Here's a shortened URL for convenience: http://j.mp/gQMjNR Michael --Original Message Text--- From: logan Date: Tue, 15 Feb 2011 21:49:26 -0800 Hello, Are there any gateways which allow me to hook a cellphone to Asterisk and use that line for routing my calls? Basically, I'm looking to play around a bit and if I can get to connect a cellphone with Asterisk then that would be great. Thanks, Hitesh PS: I have tried to search on the web, but didn't find any pointers on how to do so. -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- 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] Connect Asterisk to a cell phone
Hello, if you want use a Huawei 3G usb modem, take a look at chan_datacard module: http://wiki.e1550.mobi/doku.php Regards - Andrea-- _ -- 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] Connect Asterisk to a cell phone
On Wed, Feb 16, 2011 at 2:49 AM, logan logan...@gmail.com wrote: Hello, Are there any gateways which allow me to hook a cellphone to Asterisk and use that line for routing my calls? Basically, I'm looking to play around a bit and if I can get to connect a cellphone with Asterisk then that would be great. Thanks, Hitesh PS: I have tried to search on the web, but didn't find any pointers on how to do so. There are several pages of information here: https://wiki.asterisk.org/wiki/display/AST/Mobile+Channel ~~~ Andrew lathama Latham lath...@gmail.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
Re: [asterisk-users] Hide the plain text password (suggestion)
On Wed, Feb 16, 2011 at 12:01:20AM +0100, Hans Witvliet wrote: kept on reading the thread... Wouldn't it be better, for asterisk at least, to get rid of all this identification / authentication stuff? Keeping config files holding pain passwords or simple md5 isn't the way to solve this... Within the unix world those issues have been solved over and over again. Any chance that in 1.10 or scf we might be using something like pam? This only helps if someone has to prove the identity to you. Not if you have to prove to someone else that you know the password. In the latter case you have to actually know the plain text password, one way or the other. (If you don't, then whatever it is you know, is something a remote attacker can use). The price for using a hashes in Unix is that passwords are sent over the wire. SASL and other chalange-response authentication algorithms assume you have a common secret. And thus the server has to know the plain text password (but it is not sent in clear over the wire). -- 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
Re: [asterisk-users] how to diable echo cancellation for sip?
It is in client but not in asterisk sip channel From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 4:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to diable echo cancellation for sip? Hello, can anyboby tell me, how can I disable the echo cancellation for sip? thx a lot... best regards, Felix -- _ -- 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] function Echo() doesn't work
Did you executed Answer() before it? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 4:48 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] function Echo() doesn't work Hi guys, the function Echo() did work on CAPI, but doesn't work for SIP connection. Can anybody help? thanks a lot. best regards, Felix -- _ -- 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] Hide the plain text password
On Tue, Feb 15, 2011 at 11:51:26PM +0100, Hans Witvliet wrote: On Tue, 2011-02-15 at 07:18 -0500, Richard Kenner wrote: Anyway, the answer is: No, it's mathematically impossible to do that. Even if the passwords were stored encrypted, Asterisk itself has to be able to get the plaintext passwords to send to the remote server; so the code to decrypt them must necessarily be located on the machine. And the Source Code to Asterisk is readily available, which is how come you were able to benefit from it, so it would be trivial to extract the passwords in any case. But there IS a way to improve things, and it's what Cisco routers do. You can have all password stored in config file encrypted with a single master key. That key is stored in a special file, containing just that key. THAT file must then be heavily-protected, but all OTHER config files can now be placed into CM or anywhere else they might be needed. -- sounds like asymetric cryptography Well, it does not have to be. As I mentioned, this can already be implemented today, with #exec. And technically there's no requirement for it to use asymetric cryptography. (Now, what happens if you ever have to replace the key? The old content from the version control becomes unusable. And of course you can't keep the key in version-control) -- 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
Re: [asterisk-users] function Echo() doesn't work
Yes, I did it. The function Echo() works both on CAPI and Datacard (UMTS Stick). Just only no echo on SIP. Any suggestion? 2011/2/16 Faisal Hanif fai...@vopium.com Did you executed Answer() before it? *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Wednesday, February 16, 2011 4:48 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] function Echo() doesn't work Hi guys, the function Echo() did work on CAPI, but doesn't work for SIP connection. Can anybody help? thanks a lot. best regards, Felix -- _ -- 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] function Echo() doesn't work
Check if you have incoming SIP call in supported codec or send CLI log for further troubleshooting. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 5:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] function Echo() doesn't work Yes, I did it. The function Echo() works both on CAPI and Datacard (UMTS Stick). Just only no echo on SIP. Any suggestion? 2011/2/16 Faisal Hanif fai...@vopium.com Did you executed Answer() before it? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 4:48 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] function Echo() doesn't work Hi guys, the function Echo() did work on CAPI, but doesn't work for SIP connection. Can anybody help? thanks a lot. best regards, Felix -- _ -- 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] function Echo() doesn't work
* == Using SIP RTP CoS mark 5* *-- Executing [1174614@von-voip-provider:1] Answer(SIP/sipgate-account-, ) in new stack* *-- Executing [1174614@von-voip-provider:2] Echo(SIP/sipgate-account-, ) in new stack* * == Spawn extension (von-voip-provider, 1174614, 2) exited non-zero on 'SIP/sipgate-account-'* here is the log. It is as same as I got from CAPI and Datacard. I just didn't hear the echo from SIP connection. 2011/2/16 Faisal Hanif fai...@vopium.com Check if you have incoming SIP call in supported codec or send CLI log for further troubleshooting. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Wednesday, February 16, 2011 5:14 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] function Echo() doesn't work Yes, I did it. The function Echo() works both on CAPI and Datacard (UMTS Stick). Just only no echo on SIP. Any suggestion? 2011/2/16 Faisal Hanif fai...@vopium.com Did you executed Answer() before it? *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Wednesday, February 16, 2011 4:48 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] function Echo() doesn't work Hi guys, the function Echo() did work on CAPI, but doesn't work for SIP connection. Can anybody help? thanks a lot. best regards, Felix -- _ -- 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] How to know Caller's last position in Queue?
Hi Hanif,
I indeed use 1.8 .0 but couldn't find the channel variable for caller's
last position in queue anywhere in documentation.
Would you please let me know the channel variable name?
Thanking you.
On Wed, Feb 16, 2011 at 4:40 PM, Faisal Hanif fai...@vopium.com wrote:
If you use Asterisk 1.8.x you can have this in channel vars and can collect
and add to DB or file on h extension.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Asterisk Man
*Sent:* Wednesday, February 16, 2011 3:06 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] How to know Caller's last position in Queue?
Hi group,
I have a simple call center scenario set up on Asterisk. Customer calls the
DID and gets placed in Queue waiting for their turn to talk to the available
agent.
Sometimes Customer hangs up in between and in this case I want to get the
last position of customer in Queue.
I know there is a variable called ${QEORIGINALPOS} that gives us original
position of caller in Queue, but there doesn't seem to have something
similar for exit position.
Am I missing something?
Thanks,
--AsteriskMan
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Re: [asterisk-users] Hide the plain text password
On 02/15/2011 06:08 PM, Jian Gao wrote: How about encrypt the whole hard drive? If I built a server and give to other people, there is no easy way to stop them reset the root password or just mount my drive to read everything on it. But if build an encrypt OS then it will be secure. My question here are: 1Is this against Asterisk GPL? 2How about the performance on such a system? As long as you are providing the source code for Asterisk to anyone you distribute the binaries to, it does not matter how you distribute the binaries (encrypted or otherwise). However, encryption is not going to solve your problem: if the person you give the system to will have physical access to the system, then they will be able to access the filesystem after it is mounted. The passphrase for the filesystem has to be present at boot time for the system to be able to boot, so either it will be provided automatically or the user will be told what it is. In either case, the encryption won't end up protecting anything from the user. Encrypting filesystems or hard drives is designed to address a totally different need... it's for protecting the contents of the hard drive from someone who isn't supposed to have access to it, not the system's normal user. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] function Echo() doesn't work
I faced same issue for sipgate but got it resolved by allowing all codec in sipgate peer config. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 5:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] function Echo() doesn't work == Using SIP RTP CoS mark 5 -- Executing [1174614@von-voip-provider:1] Answer(SIP/sipgate-account-, ) in new stack -- Executing [1174614@von-voip-provider:2] Echo(SIP/sipgate-account-, ) in new stack == Spawn extension (von-voip-provider, 1174614, 2) exited non-zero on 'SIP/sipgate-account-' here is the log. It is as same as I got from CAPI and Datacard. I just didn't hear the echo from SIP connection. 2011/2/16 Faisal Hanif fai...@vopium.com Check if you have incoming SIP call in supported codec or send CLI log for further troubleshooting. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 5:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] function Echo() doesn't work Yes, I did it. The function Echo() works both on CAPI and Datacard (UMTS Stick). Just only no echo on SIP. Any suggestion? 2011/2/16 Faisal Hanif fai...@vopium.com Did you executed Answer() before it? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 4:48 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] function Echo() doesn't work Hi guys, the function Echo() did work on CAPI, but doesn't work for SIP connection. Can anybody help? thanks a lot. best regards, Felix -- _ -- 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] function Echo() doesn't work
In which conf-Data should I allow all codec? Thank u for explaining. 2011/2/16 Faisal Hanif fai...@vopium.com I faced same issue for sipgate but got it resolved by allowing all codec in sipgate peer config. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Wednesday, February 16, 2011 5:33 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] function Echo() doesn't work * == Using SIP RTP CoS mark 5* *-- Executing [1174614@von-voip-provider:1] Answer(SIP/sipgate-account-, ) in new stack* *-- Executing [1174614@von-voip-provider:2] Echo(SIP/sipgate-account-, ) in new stack* * == Spawn extension (von-voip-provider, 1174614, 2) exited non-zero on 'SIP/sipgate-account-'* here is the log. It is as same as I got from CAPI and Datacard. I just didn't hear the echo from SIP connection. 2011/2/16 Faisal Hanif fai...@vopium.com Check if you have incoming SIP call in supported codec or send CLI log for further troubleshooting. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Wednesday, February 16, 2011 5:14 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] function Echo() doesn't work Yes, I did it. The function Echo() works both on CAPI and Datacard (UMTS Stick). Just only no echo on SIP. Any suggestion? 2011/2/16 Faisal Hanif fai...@vopium.com Did you executed Answer() before it? *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Wednesday, February 16, 2011 4:48 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] function Echo() doesn't work Hi guys, the function Echo() did work on CAPI, but doesn't work for SIP connection. Can anybody help? thanks a lot. best regards, Felix -- _ -- 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 -- _ -- 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 know Caller's last position in Queue?
You have enable following in queue configuration,
setinterfacevar=yes
setqueueentryvar=yes
setqueuevar=yes
and you will find your data in following variables,
${QEORIGINALPOS} will have position when caller enter the queue.
${QUEUEPOSITION} will have position when caller left the queue.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Man
Sent: Wednesday, February 16, 2011 5:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to know Caller's last position in Queue?
Hi Hanif,
I indeed use 1.8 .0 but couldn't find the channel variable for caller's
last position in queue anywhere in documentation.
Would you please let me know the channel variable name?
Thanking you.
On Wed, Feb 16, 2011 at 4:40 PM, Faisal Hanif fai...@vopium.com wrote:
If you use Asterisk 1.8.x you can have this in channel vars and can collect
and add to DB or file on h extension.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Man
Sent: Wednesday, February 16, 2011 3:06 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to know Caller's last position in Queue?
Hi group,
I have a simple call center scenario set up on Asterisk. Customer calls the
DID and gets placed in Queue waiting for their turn to talk to the available
agent.
Sometimes Customer hangs up in between and in this case I want to get the
last position of customer in Queue.
I know there is a variable called ${QEORIGINALPOS} that gives us original
position of caller in Queue, but there doesn't seem to have something
similar for exit position.
Am I missing something?
Thanks,
--AsteriskMan
--
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Re: [asterisk-users] function Echo() doesn't work
I tried to set allow=all in sip.conf. But it still doesn't work. 2011/2/16 Faisal Hanif fai...@vopium.com I faced same issue for sipgate but got it resolved by allowing all codec in sipgate peer config. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Wednesday, February 16, 2011 5:33 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] function Echo() doesn't work * == Using SIP RTP CoS mark 5* *-- Executing [1174614@von-voip-provider:1] Answer(SIP/sipgate-account-, ) in new stack* *-- Executing [1174614@von-voip-provider:2] Echo(SIP/sipgate-account-, ) in new stack* * == Spawn extension (von-voip-provider, 1174614, 2) exited non-zero on 'SIP/sipgate-account-'* here is the log. It is as same as I got from CAPI and Datacard. I just didn't hear the echo from SIP connection. 2011/2/16 Faisal Hanif fai...@vopium.com Check if you have incoming SIP call in supported codec or send CLI log for further troubleshooting. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Wednesday, February 16, 2011 5:14 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] function Echo() doesn't work Yes, I did it. The function Echo() works both on CAPI and Datacard (UMTS Stick). Just only no echo on SIP. Any suggestion? 2011/2/16 Faisal Hanif fai...@vopium.com Did you executed Answer() before it? *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Wednesday, February 16, 2011 4:48 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] function Echo() doesn't work Hi guys, the function Echo() did work on CAPI, but doesn't work for SIP connection. Can anybody help? thanks a lot. best regards, Felix -- _ -- 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 -- _ -- 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] Hide the plain text password
ken...@gnat.com (Richard Kenner) writes:
Here's a possible design:
- There's optionally a file in the config
directory called master_key. It contains just a string.
- A CLI command core encrypt string is added to Asterisk. It takes the
provided string, encrypts it using the string in master_key, and outputs
a string of the form {enc:encrypted_version_of_string}.
- The config file reader looks for strings of the form {enc:string}:
and replaces them, before otherwise parsing the line, with the decrypted
version of the string using the key in the master_key file.
This sounds pretty reasonable, except perhaps that you might only want
to convert strings in password fields -- otherwise you risk false
positives in e.g. the dial plan.
I can recommend contracting with one of the indepedent Asterisk
developers to get this done. You will likely find them on the
Asterisk-biz-list.
/Benny
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Re: [asterisk-users] function Echo() doesn't work
Did you make any peer for sipgate if yes then do for that peers. Please also note that disallow line should be before allow line. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 6:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] function Echo() doesn't work I tried to set allow=all in sip.conf. But it still doesn't work. 2011/2/16 Faisal Hanif fai...@vopium.com I faced same issue for sipgate but got it resolved by allowing all codec in sipgate peer config. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 5:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] function Echo() doesn't work == Using SIP RTP CoS mark 5 -- Executing [1174614@von-voip-provider:1] Answer(SIP/sipgate-account-, ) in new stack -- Executing [1174614@von-voip-provider:2] Echo(SIP/sipgate-account-, ) in new stack == Spawn extension (von-voip-provider, 1174614, 2) exited non-zero on 'SIP/sipgate-account-' here is the log. It is as same as I got from CAPI and Datacard. I just didn't hear the echo from SIP connection. 2011/2/16 Faisal Hanif fai...@vopium.com Check if you have incoming SIP call in supported codec or send CLI log for further troubleshooting. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 5:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] function Echo() doesn't work Yes, I did it. The function Echo() works both on CAPI and Datacard (UMTS Stick). Just only no echo on SIP. Any suggestion? 2011/2/16 Faisal Hanif fai...@vopium.com Did you executed Answer() before it? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 4:48 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] function Echo() doesn't work Hi guys, the function Echo() did work on CAPI, but doesn't work for SIP connection. Can anybody help? thanks a lot. best regards, Felix -- _ -- 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 -- _ -- 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] Hide the plain text password
- The config file reader looks for strings of the form {enc:string}:
and replaces them, before otherwise parsing the line, with the decrypted
version of the string using the key in the master_key file.
This sounds pretty reasonable, except perhaps that you might only want
to convert strings in password fields -- otherwise you risk false
positives in e.g. the dial plan.
I think this works much better if it's purely lexical. Otherwise, you
have to teach the code what's a password and what's not and maintaning
that is an ongoing issue, so I think a cleaner design would be to pick
some string that's just not going to occur anywhere.
I can recommend contracting with one of the indepedent Asterisk
developers to get this done. You will likely find them on the
Asterisk-biz-list.
I could easily do it myself if it were something that I personally needed
(except that I'm not sure if two-way encryption routines already exist
in Asterisk), but we don't have enough passwords for this to be an issue.
I was posting the design to address the issues raised by the person who
started the thread.
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[asterisk-users] Play one audio file to the called part before the Dial() command
Hi, I am not sure if it is doable: 1. We originate one call from Asterisk 2. Asterisk plays one audio file to the called part when the called part picks up the phone. 3. Asterisk establish one real connection between the caller part and the called part. Thanks, Songtao Yu-- _ -- 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] function Echo() doesn't work
Yes, I did it exactly as what you said. It still doesn't work. :-( 2011/2/16 Faisal Hanif fai...@vopium.com Did you make any peer for sipgate if yes then do for that peers. Please also note that disallow line should be before allow line. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Wednesday, February 16, 2011 6:22 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] function Echo() doesn't work I tried to set allow=all in sip.conf. But it still doesn't work. 2011/2/16 Faisal Hanif fai...@vopium.com I faced same issue for sipgate but got it resolved by allowing all codec in sipgate peer config. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Wednesday, February 16, 2011 5:33 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] function Echo() doesn't work * == Using SIP RTP CoS mark 5* *-- Executing [1174614@von-voip-provider:1] Answer(SIP/sipgate-account-, ) in new stack* *-- Executing [1174614@von-voip-provider:2] Echo(SIP/sipgate-account-, ) in new stack* * == Spawn extension (von-voip-provider, 1174614, 2) exited non-zero on 'SIP/sipgate-account-'* here is the log. It is as same as I got from CAPI and Datacard. I just didn't hear the echo from SIP connection. 2011/2/16 Faisal Hanif fai...@vopium.com Check if you have incoming SIP call in supported codec or send CLI log for further troubleshooting. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Wednesday, February 16, 2011 5:14 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] function Echo() doesn't work Yes, I did it. The function Echo() works both on CAPI and Datacard (UMTS Stick). Just only no echo on SIP. Any suggestion? 2011/2/16 Faisal Hanif fai...@vopium.com Did you executed Answer() before it? *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Wednesday, February 16, 2011 4:48 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] function Echo() doesn't work Hi guys, the function Echo() did work on CAPI, but doesn't work for SIP connection. Can anybody help? thanks a lot. best regards, Felix -- _ -- 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 -- _ -- 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] Play one audio file to the called part before the Dial() command
You can do it using callback files or AMI. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Songtao Yu Sent: Wednesday, February 16, 2011 6:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Play one audio file to the called part before the Dial() command Hi, I am not sure if it is doable: 1. We originate one call from Asterisk 2. Asterisk plays one audio file to the called part when the called part picks up the phone. 3. Asterisk establish one real connection between the caller part and the called part. Thanks, Songtao Yu -- _ -- 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] DTMF not detected, time out
It is somehow back to normal. Nothing change. May be the sip provider problem. However, it lasts for quite a while. Thanks On Wed, Feb 16, 2011 at 12:04 PM, Faisal Hanif fai...@vopium.com wrote: You can also append add dtmf logging to cosole and see if dtmf is coming from carrier. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *asterisk asterisk *Sent:* Wednesday, February 16, 2011 8:58 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] DTMF not detected, time out In the past it was set as auto and worked. I change to RFC2833 but did not work. How can I check further? On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif fai...@vopium.com wrote: Check if dtmfmode is properly set on SIP trunk ask with your carrier which dmtfmode they support. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *asterisk asterisk *Sent:* Wednesday, February 16, 2011 5:39 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] DTMF not detected, time out Hi, I encounter this problem recently after quite some months of my asterisk. I have a SIP trunk for dial in and out. When dial-in, it matches the callerid number and decides. If matched, it will either go into an a very brief IVR. The IVR allows caller to dial internal extension. All along it is working well both from outside call and internal users. Now for unknown reason, it fails with a timeout and hangup. It is the only message I can see at the console. But internal user can do this without any problem. Appreciate if someone can help. CK -- _ -- 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] DTMF not detected, time out
In your sip.conf, in trunk parameters use: dtmfmode = INFO Date: Wed, 16 Feb 2011 23:07:16 +0800 From: aster...@ck-lee.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DTMF not detected, time out It is somehow back to normal. Nothing change. May be the sip provider problem. However, it lasts for quite a while. Thanks On Wed, Feb 16, 2011 at 12:04 PM, Faisal Hanif fai...@vopium.com wrote: You can also append add dtmf logging to cosole and see if dtmf is coming from carrier. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk Sent: Wednesday, February 16, 2011 8:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF not detected, time out In the past it was set as auto and worked. I change to RFC2833 but did not work. How can I check further? On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif fai...@vopium.com wrote:Check if dtmfmode is properly set on SIP trunk ask with your carrier which dmtfmode they support. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk Sent: Wednesday, February 16, 2011 5:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DTMF not detected, time out Hi, I encounter this problem recently after quite some months of my asterisk. I have a SIP trunk for dial in and out. When dial-in, it matches the callerid number and decides. If matched, it will either go into an a very brief IVR. The IVR allows caller to dial internal extension. All along it is working well both from outside call and internal users. Now for unknown reason, it fails with a timeout and hangup. It is the only message I can see at the console. But internal user can do this without any problem. Appreciate if someone can help. CK -- _ -- 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 -- _ -- 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] function Echo() doesn't work
Hi Felix, Am Mittwoch, den 16.02.2011, 12:47 +0100 schrieb Felix Dong: Hi guys, the function Echo() did work on CAPI, but doesn't work for SIP connection. Can anybody help? thanks a lot. are You trying to echo between local phones or is it a external call via some VoIP Provider? In latter case: do You forward the RTP traffic from external to Your asterisk? The relevant ports can be configured in rtp.conf. Configure at least 4 ports per connection. Configure port forwarding for this range of UDP ports in Youe NAT-device (e.g. router or firewall). HTH, Karsten best regards, Felix -- _ -- 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] Connect Asterisk to a cell phone
Hello all, Thank you for the responses. I really appreciate it. Since I'm just trying it out for fun, I will begin by using mobile-chan and see how that goes. Thanks a lot, Hitesh On Wed, Feb 16, 2011 at 4:05 AM, Andrew Latham lath...@gmail.com wrote: On Wed, Feb 16, 2011 at 2:49 AM, logan logan...@gmail.com wrote: Hello, Are there any gateways which allow me to hook a cellphone to Asterisk and use that line for routing my calls? Basically, I'm looking to play around a bit and if I can get to connect a cellphone with Asterisk then that would be great. Thanks, Hitesh PS: I have tried to search on the web, but didn't find any pointers on how to do so. There are several pages of information here: https://wiki.asterisk.org/wiki/display/AST/Mobile+Channel ~~~ Andrew lathama Latham lath...@gmail.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
[asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP
How should I configure my asterisk server so that I can receive calls from an unregistered peer from whom I also receive registrations of sip phones? I'm asking you this, because with my actual configuration, when I register a contact from that peer's IP, no more inbound calls are accepted from that peer, as my asterisk rejects those INVITEs with 407 Proxy Authentication Required, I assume because they don't carry the registered contact registration!!! My SIP contacts have type=friend and all inbound calls not coming from my registered phones fall in the default context without authentication, so that someone in the Internet be able to call freely through the Internet anyone in my server's dial plan. Some ideas? Regards, Ricardo Carvalho. -- _ -- 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] trunk not working if I register a phone at the same IP as the trunk peer's IP
Well a quick n easy fix for you is you can configure you call sending peers to use username secret in INVITE. As far as I know it possible in almost all CISCO, Avaya and all other standard Gateway and SBCs which follows full SIP RFCs. If you can't do it then you need to use curl as realtime engine instead of MySQL. It will call a URL for each SIP request which you can handle with flexibility in your CGI scripts with apache. But be careful as per my experience asterisk 1.6 with curl as realtime engine can handle a max of 120 registration in parallel if registration refresh time is 120 seconds. Faisal Hanif From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ricardo Carvalho Sent: Wednesday, February 16, 2011 9:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP How should I configure my asterisk server so that I can receive calls from an unregistered peer from whom I also receive registrations of sip phones? I'm asking you this, because with my actual configuration, when I register a contact from that peer's IP, no more inbound calls are accepted from that peer, as my asterisk rejects those INVITEs with 407 Proxy Authentication Required, I assume because they don't carry the registered contact registration!!! My SIP contacts have type=friend and all inbound calls not coming from my registered phones fall in the default context without authentication, so that someone in the Internet be able to call freely through the Internet anyone in my server's dial plan. Some ideas? Regards, Ricardo Carvalho. -- _ -- 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] Lua extensions are not working on asterisk 1.8.2.3
I was messing with something in conf dir. I reinstalled asterisk and removed extensions.conf and lua extensions is working now. I think lua in dialplan is a killer feature. It enables complex apps to be done in a much easier way now. 2011/2/16 Faisal Hanif fai...@vopium.com: You may need to share your LUA code and the extension your call is need to execute. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlo Pires Sent: Wednesday, February 16, 2011 3:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Lua extensions are not working on asterisk 1.8.2.3 But when I try to call one extension created with lua I got a message telling that extension doesnt exist on default context. Am I missing something? 2011/2/15 Tilghman Lesher tilgh...@meg.abyt.es: On Tuesday 15 February 2011 11:06:32 Carlo Pires wrote: Hi, After compiling a installing asterisk 1.8.2.3 I wanted to play with lua but I noticed that extensions created in extensions.lua was not being registered with asterisk. uga1*CLI dialplan show [ Context 'app_queue_gosub_virtual_context' created by 'app_queue' ] 's' = 1. NoOp() [app_queue] [ Context 'parkedcalls' created by 'features' ] '700' = 1. Park() [features] [ Context 'app_dial_gosub_virtual_context' created by 'app_dial' ] 's' = 1. NoOp() [app_dial] [ Context 'local' created by 'pbx_lua' ] Alt. Switch = 'Lua/' [pbx_lua] [ Context 'demo' created by 'pbx_lua' ] Alt. Switch = 'Lua/' [pbx_lua] [ Context 'default' created by 'pbx_lua' ] Alt. Switch = 'Lua/' [pbx_lua] -= 3 extensions (3 priorities) in 6 contexts. =- uga1*CLI uga1*CLI dialplan show demo [ Context 'demo' created by 'pbx_lua' ] Alt. Switch = 'Lua/' [pbx_lua] -= 0 extensions (0 priorities) in 1 context. =- uga1*CLI Need I enable something to get lua extensions to be created? No, that's how Lua extensions work, with the switch statement. Your extensions are still being evaluated by Lua. The only difference is that pbx_lua now doesn't see any need to create extensions, because it will see every extension when it hits the switch. -- Tilghman -- _ -- 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 -- _ -- 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] trunk not working if I register a phone at the same IP as the trunk peer's IP
Isn't this a limitation that can be surpassed with some configuration that I'm lacking in my sip.conf or extensions.conf of my asterisk? Ricardo. On Wed, Feb 16, 2011 at 4:54 PM, Faisal Hanif fai...@vopium.com wrote: Well a quick n easy fix for you is you can configure you call sending peers to use username secret in INVITE. As far as I know it possible in almost all CISCO, Avaya and all other standard Gateway and SBCs which follows full SIP RFCs. If you can’t do it then you need to use curl as realtime engine instead of MySQL. It will call a URL for each SIP request which you can handle with flexibility in your CGI scripts with apache. But be careful as per my experience asterisk 1.6 with curl as realtime engine can handle a max of 120 registration in parallel if registration refresh time is 120 seconds. Faisal Hanif *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ricardo Carvalho *Sent:* Wednesday, February 16, 2011 9:41 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP How should I configure my asterisk server so that I can receive calls from an unregistered peer from whom I also receive registrations of sip phones? I'm asking you this, because with my actual configuration, when I register a contact from that peer's IP, no more inbound calls are accepted from that peer, as my asterisk rejects those INVITEs with 407 Proxy Authentication Required, I assume because they don't carry the registered contact registration!!! My SIP contacts have type=friend and all inbound calls not coming from my registered phones fall in the default context without authentication, so that someone in the Internet be able to call freely through the Internet anyone in my server's dial plan. Some ideas? Regards, Ricardo Carvalho. -- _ -- 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] pipe audio stream to external application
Hi, I'd like to know if there's an easy way of doing the following: SIP phone dials a custom feature code in Asterisk, call gets answered within a custom context (Answer()), anything that the caller says should be redirected/piped to an external application. Something like monitor except audio should be sent live. More like app_ices (or app_ezstream if that existed) but for a generic app. Thanks Vieri -- _ -- 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] Aastra phones cannot transfer calls?
On 16 February 2011 00:22, Ernie Dunbar maill...@lightspeed.ca wrote: At 12:12 PM 2/15/2011, you wrote: I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk v1.6.2.1. They can call each other's extensions (and make and receive calls otherwise), but they cannot transfer calls, not even to outside I'm running 1.6.2.16.1 and have three Aastra 480i phones and have had no problem at all with transfers. Have you considered trying a newer version? Nope. Upgraded to 1.6.2.16.1, and I still see the same effect. It may be a setting on the phone or a SIP setting. I'll investigate this elsewhere but report back about the solution. I also tried this with a 6757i and a 6753i with no problems (blind and attended) on Asterisk 1.6.2.16.1. Have you updated the handset firmware to 2.6.0.2010? What SIP settings do you have in Asterisk? -- _ -- 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] Asterisk on a USB with persistence
Hi, I'm looking to get an ISO of FreePBX or AsteriskNOW installed on a USB that I can boot from and also be able to save my changes. Is this possible? My search on web doesn't seem to find anything useful. For now I don't have the option of having a spare machine or creating a partition on my existing one for my experiments with Asterisk. My end goal is to have chan_mobile configured and see if I can make calls through my cellphone using that. Thanks, Hitesh -- _ -- 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] trunk not working if I register a phone at the same IP as the trunk peer's IP
I have played a lot on this issue with asterisk config but in realtime it doesn't supported static peers with version 1.6.2.14. From: Ricardo Carvalho [mailto:rjcarvalho.li...@gmail.com] Sent: Wednesday, February 16, 2011 10:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Faisal Hanif Subject: Re: [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP Isn't this a limitation that can be surpassed with some configuration that I'm lacking in my sip.conf or extensions.conf of my asterisk? Ricardo. On Wed, Feb 16, 2011 at 4:54 PM, Faisal Hanif fai...@vopium.com wrote: Well a quick n easy fix for you is you can configure you call sending peers to use username secret in INVITE. As far as I know it possible in almost all CISCO, Avaya and all other standard Gateway and SBCs which follows full SIP RFCs. If you can't do it then you need to use curl as realtime engine instead of MySQL. It will call a URL for each SIP request which you can handle with flexibility in your CGI scripts with apache. But be careful as per my experience asterisk 1.6 with curl as realtime engine can handle a max of 120 registration in parallel if registration refresh time is 120 seconds. Faisal Hanif From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ricardo Carvalho Sent: Wednesday, February 16, 2011 9:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP How should I configure my asterisk server so that I can receive calls from an unregistered peer from whom I also receive registrations of sip phones? I'm asking you this, because with my actual configuration, when I register a contact from that peer's IP, no more inbound calls are accepted from that peer, as my asterisk rejects those INVITEs with 407 Proxy Authentication Required, I assume because they don't carry the registered contact registration!!! My SIP contacts have type=friend and all inbound calls not coming from my registered phones fall in the default context without authentication, so that someone in the Internet be able to call freely through the Internet anyone in my server's dial plan. Some ideas? Regards, Ricardo Carvalho. -- _ -- 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] pipe audio stream to external application
EAGI could be your target application. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Wednesday, February 16, 2011 11:01 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] pipe audio stream to external application Hi, I'd like to know if there's an easy way of doing the following: SIP phone dials a custom feature code in Asterisk, call gets answered within a custom context (Answer()), anything that the caller says should be redirected/piped to an external application. Something like monitor except audio should be sent live. More like app_ices (or app_ezstream if that existed) but for a generic app. Thanks Vieri -- _ -- 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] Asterisk on a USB with persistence
You can simply use Portable LinuxLive USB Creator 2.6 or grub4dos. And make your USB bootable by any Linux Live ISO. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of logan Sent: Wednesday, February 16, 2011 11:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk on a USB with persistence Hi, I'm looking to get an ISO of FreePBX or AsteriskNOW installed on a USB that I can boot from and also be able to save my changes. Is this possible? My search on web doesn't seem to find anything useful. For now I don't have the option of having a spare machine or creating a partition on my existing one for my experiments with Asterisk. My end goal is to have chan_mobile configured and see if I can make calls through my cellphone using that. Thanks, Hitesh -- _ -- 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] Aastra phones cannot transfer calls?
On 16 February 2011 00:22, Ernie Dunbar maill...@lightspeed.ca wrote: At 12:12 PM 2/15/2011, you wrote: I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk v1.6.2.1. They can call each other's extensions (and make and receive calls otherwise), but they cannot transfer calls, not even to outside I'm running 1.6.2.16.1 and have three Aastra 480i phones and have had no problem at all with transfers. Have you considered trying a newer version? Nope. Upgraded to 1.6.2.16.1, and I still see the same effect. It may be a setting on the phone or a SIP setting. I'll investigate this elsewhere but report back about the solution. I also tried this with a 6757i and a 6753i with no problems (blind and attended) on Asterisk 1.6.2.16.1. Have you updated the handset firmware to 2.6.0.2010? What SIP settings do you have in Asterisk? Actually, I found the problem. Allowtransfer is a new SIP setting (it certainly isn't in sip.conf on the old server) and by default it's set to no globally. Changing this to yes fixes the problem. -- _ -- 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] Polycom IP335
I am posting here since you guys are my last hope. I am trying to configure a Polycom Soundpoint IP 335 with MWI. Is there any way to eliminate the scrolling messages and Msgs softkey? I am trying to get it where it's just the light that indicates the new messages. I don't know if Asterisk has to send a different notification or what have you. Thanks, --Eric -- _ -- 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 IP335
On Wed, Feb 16, 2011 at 4:51 PM, ERIC HERRON e...@lanline.com wrote: I am posting here since you guys are my last hope. I am trying to configure a Polycom Soundpoint IP 335 with MWI. Is there any way to eliminate the scrolling messages and Msgs softkey? I am trying to get it where it’s just the light that indicates the new messages. I don’t know if Asterisk has to send a different notification or what have you. Thanks, --Eric http://supportdocs.polycom.com/PolycomService/support/global/documents/support/setup_maintenance/products/voice/spip_ssip_Admin_Guide_UCS_v3_3_0.pdf ~~~ Andrew lathama Latham lath...@gmail.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
Re: [asterisk-users] Polycom IP335
What I am trying to achieve is not in there. Thanks though. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham Sent: Wednesday, February 16, 2011 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom IP335 On Wed, Feb 16, 2011 at 4:51 PM, ERIC HERRON e...@lanline.com wrote: I am posting here since you guys are my last hope. I am trying to configure a Polycom Soundpoint IP 335 with MWI. Is there any way to eliminate the scrolling messages and Msgs softkey? I am trying to get it where it’s just the light that indicates the new messages. I don’t know if Asterisk has to send a different notification or what have you. Thanks, --Eric http://supportdocs.polycom.com/PolycomService/support/global/documents/support/setup_maintenance/products/voice/spip_ssip_Admin_Guide_UCS_v3_3_0.pdf ~~~ Andrew lathama Latham lath...@gmail.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 _ No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1204 / Virus Database: 1435/3447 - Release Date: 02/16/11 -- _ -- 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 IP335
On Wed, Feb 16, 2011 at 2:51 PM, ERIC HERRON e...@lanline.com wrote: I am posting here since you guys are my last hope. I am trying to configure a Polycom Soundpoint IP 335 with MWI. Is there any way to eliminate the scrolling messages and Msgs softkey? I am trying to get it where it’s just the light that indicates the new messages. I don’t know if Asterisk has to send a different notification or what have you. Thanks, --Eric I've had that same request a few times. I've looked through the Polycom manual, even the new UC software 3.3.1, and never found the setting for it. It is either all or nothing for MWI. The scrolling messages is the part I get complaints about. People would rather have the clock shown on the screen. Ryan -- _ -- 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 IP335
I have it on the 430s. I think it's a firmware issue but I am having trouble replicating it on the 430 I could have sworn I had it on one phone before I rebooted it but memory might be influenced by hopes. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner Sent: Wednesday, February 16, 2011 3:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom IP335 On Wed, Feb 16, 2011 at 2:51 PM, ERIC HERRON e...@lanline.com wrote: I am posting here since you guys are my last hope. I am trying to configure a Polycom Soundpoint IP 335 with MWI. Is there any way to eliminate the scrolling messages and Msgs softkey? I am trying to get it where it's just the light that indicates the new messages. I don't know if Asterisk has to send a different notification or what have you. Thanks, --Eric I've had that same request a few times. I've looked through the Polycom manual, even the new UC software 3.3.1, and never found the setting for it. It is either all or nothing for MWI. The scrolling messages is the part I get complaints about. People would rather have the clock shown on the screen. Ryan -- _ -- 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 _ No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1204 / Virus Database: 1435/3447 - Release Date: 02/16/11 -- _ -- 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 IP335
On Wed, Feb 16, 2011 at 3:05 PM, ERIC HERRON e...@lanline.com wrote: I have it on the 430s. I think it’s a firmware issue but I am having trouble replicating it on the 430 I could have sworn I had it on one phone before I rebooted it but memory might be influenced by hopes. What setting were you using to configure it that way. I've was running 3.2.3 and am now using 3.3.1 on the IP335s and never had luck disabling the scrolling message. Ryan -- _ -- 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 ring tone on inbound call - but channel connects fine
Hi Everyone, I have a SIP turnk which works fine with both inbound and outbound calling. However, the only issue is that there is no Ring Tone if someone calls us. The phones used are Aastra and Polycom connected to the PBX via VPN (SIP). I do get an outbound ring tone, so it's not that there is any media loss between the phones and the PBX. But when the DID is called there is dead silence until the call is picked up. What is generally causing something like this? and where should I start looking? Much appreciate your experienced tips. Thanks -- _ -- 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] Cisco 7945G phone with asterisk
dear I have a good exp in setting up 79xx on sccp, with sccp-b library, and tftp server, which part is the main problem for you? best On Wed, Feb 16, 2011 at 3:10 PM, Andrew Latham lath...@gmail.com wrote: On Wed, Feb 16, 2011 at 7:32 AM, ast guy ast...@gmail.com wrote: Hi, Anyone who has deployed Cisco 7945G phone with asterisk, kindly share your experience. /ag I did some 7941's a few months ago with SIP. They work pretty well. Make a console cable for the AUX port and you can see them load. I had to add Spanish menus to them so I ended up hacking the load process (Cisco does not support language files with SIP firmware). The 7945 should just have more features. ~~~ Andrew lathama Latham lath...@gmail.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
Re: [asterisk-users] Polycom IP335
On IP430s cat sip.ver VVX-1500 3.2.2.0481 All others 3.2.2.0477 2345-11402-001.bootrom.ld sip.ld Phone1.cfg msg msg.bypassInstantMessage=1 mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=*97 msg.mwi.2.subscribe= Sip.cfg up.mwiVisible=0 Is there anywhere else to look? Its bothering me to no end. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner Sent: Wednesday, February 16, 2011 3:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom IP335 On Wed, Feb 16, 2011 at 3:05 PM, ERIC HERRON e...@lanline.com wrote: I have it on the 430s. I think it's a firmware issue but I am having trouble replicating it on the 430 I could have sworn I had it on one phone before I rebooted it but memory might be influenced by hopes. What setting were you using to configure it that way. I've was running 3.2.3 and am now using 3.3.1 on the IP335s and never had luck disabling the scrolling message. Ryan -- _ -- 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 _ No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1204 / Virus Database: 1435/3447 - Release Date: 02/16/11 -- _ -- 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] DTMF not detected, time out
some outside sip provider does not accept dtmf, if you have not this problem in your local, ask your outside carrier best On Wed, Feb 16, 2011 at 7:27 AM, asterisk asterisk aster...@ck-lee.comwrote: In the past it was set as auto and worked. I change to RFC2833 but did not work. How can I check further? On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif fai...@vopium.com wrote: Check if dtmfmode is properly set on SIP trunk ask with your carrier which dmtfmode they support. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *asterisk asterisk *Sent:* Wednesday, February 16, 2011 5:39 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] DTMF not detected, time out Hi, I encounter this problem recently after quite some months of my asterisk. I have a SIP trunk for dial in and out. When dial-in, it matches the callerid number and decides. If matched, it will either go into an a very brief IVR. The IVR allows caller to dial internal extension. All along it is working well both from outside call and internal users. Now for unknown reason, it fails with a timeout and hangup. It is the only message I can see at the console. But internal user can do this without any problem. Appreciate if someone can help. CK -- _ -- 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] No ring tone on inbound call - but channelconnects fine
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Wednesday, February 16, 2011 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] No ring tone on inbound call - but channelconnects fine Hi Everyone, I have a SIP turnk which works fine with both inbound and outbound calling. However, the only issue is that there is no Ring Tone if someone calls us. The phones used are Aastra and Polycom connected to the PBX via VPN (SIP). I do get an outbound ring tone, so it's not that there is any media loss between the phones and the PBX. But when the DID is called there is dead silence until the call is picked up. What is generally causing something like this? and where should I start looking? Much appreciate your experienced tips. Thanks This sounds like a dialplan problem. My thought is that your SIP trunk should go to an incoming context that does something like this: In-house phones are 1000 and 1001 [incoming] Exten = s,1,answer Exten = s,n,Dial(SIP/1000SIP/1001,30,mKkTt) -- _ -- 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] No ring tone on inbound call - but channelconnects fine
Thanks. Indeed ringing instead of MoH which was missing files fixed the issue. Thanks for the quick great tip. Simple things hide from us sometime. On Wed, Feb 16, 2011 at 3:48 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B *Sent:* Wednesday, February 16, 2011 2:33 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] No ring tone on inbound call - but channelconnects fine Hi Everyone, I have a SIP turnk which works fine with both inbound and outbound calling. However, the only issue is that there is no Ring Tone if someone calls us. The phones used are Aastra and Polycom connected to the PBX via VPN (SIP). I do get an outbound ring tone, so it's not that there is any media loss between the phones and the PBX. But when the DID is called there is dead silence until the call is picked up. What is generally causing something like this? and where should I start looking? Much appreciate your experienced tips. Thanks This sounds like a dialplan problem. My thought is that your SIP trunk should go to an “incoming” context that does something like this: In-house phones are 1000 and 1001 [incoming] Exten = s,1,answer Exten = s,n,Dial(SIP/1000SIP/1001,30,mKkTt) -- _ -- 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] Polycom IP335
On 02/16/2011 09:43 PM, ERIC HERRON wrote: On IP430s cat sip.ver VVX-1500 3.2.2.0481 All others 3.2.2.0477 2345-11402-001.bootrom.ld sip.ld Phone1.cfg msg msg.bypassInstantMessage=1 mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=*97 msg.mwi.2.subscribe= Sip.cfg up.mwiVisible=0 Is there anywhere else to look? Its bothering me to no end. I share your pain. I have an IP335 and IP670 here. Have not configured the IP335 yet but using the latest Admin Guide (3.3.1) did configure the IP670 running the latest bootrom (4.3.0) and firmware (3.3.1). Problems on the IP670: 1) it seems impossible to turn off the backlight 2) it seems impossible to disable that stupid periodic MWI sound. Whoever at Polycom thought that that was a good idea should meet a seriously big clue-by-4. To me it seems like their 3.3.x branch could use a few bugfixes... Have you tried an older or newer release? Regards, Patrick -- _ -- 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 IP335
On Wed, Feb 16, 2011 at 5:49 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: I share your pain. I have an IP335 and IP670 here. Have not configured the IP335 yet but using the latest Admin Guide (3.3.1) did configure the IP670 running the latest bootrom (4.3.0) and firmware (3.3.1). Problems on the IP670: 1) it seems impossible to turn off the backlight 2) it seems impossible to disable that stupid periodic MWI sound. Whoever at Polycom thought that that was a good idea should meet a seriously big clue-by-4. To me it seems like their 3.3.x branch could use a few bugfixes... Have you tried an older or newer release? Regards, Patrick Backlight works fine on a IP550 with 3.3.1 . I have mine set to off when idle. I like that the 3.3.x series doesn't required the default sip.cfg and phone1.cfg files. The structure of the XML seems cleaner and more consistent. up up.backlight up.backlight.idleIntensity=0 up.backlight.onIntensity=3 /up.backlight /up The only bug I've seen with 3.3.1 is on the IP335. After dialing when it connects the caller name and number jump 1 pixel higher, which looks weird as it is close to the line. One 3.2.3 it didn't move up and looked centered. However the scrolling caller id for incoming calls make this minor annoyance worth the upgrade. Ryan -- _ -- 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 IP335
I haven't played with the backlights yet. One annoyance at a time. To disabled the mwi chirp can be set to silence. MESSAGE_WAITING se.pat.misc.1.name=message waiting se.pat.misc.1.inst.1.type=silence .. I am trying the different firmwares now to see if it makes any difference. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner Sent: Wednesday, February 16, 2011 6:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom IP335 On Wed, Feb 16, 2011 at 5:49 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: I share your pain. I have an IP335 and IP670 here. Have not configured the IP335 yet but using the latest Admin Guide (3.3.1) did configure the IP670 running the latest bootrom (4.3.0) and firmware (3.3.1). Problems on the IP670: 1) it seems impossible to turn off the backlight 2) it seems impossible to disable that stupid periodic MWI sound. Whoever at Polycom thought that that was a good idea should meet a seriously big clue-by-4. To me it seems like their 3.3.x branch could use a few bugfixes... Have you tried an older or newer release? Regards, Patrick Backlight works fine on a IP550 with 3.3.1 . I have mine set to off when idle. I like that the 3.3.x series doesn't required the default sip.cfg and phone1.cfg files. The structure of the XML seems cleaner and more consistent. up up.backlight up.backlight.idleIntensity=0 up.backlight.onIntensity=3 /up.backlight /up The only bug I've seen with 3.3.1 is on the IP335. After dialing when it connects the caller name and number jump 1 pixel higher, which looks weird as it is close to the line. One 3.2.3 it didn't move up and looked centered. However the scrolling caller id for incoming calls make this minor annoyance worth the upgrade. Ryan -- _ -- 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 _ No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1204 / Virus Database: 1435/3447 - Release Date: 02/16/11 -- _ -- 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] CDR with unix time.
What module are you using? I have the cdr_mysql_addon.so module, and I can define alias for the collums. I have an alias for start and end variable, and they both get recorded as Unix EPOCH integers values on mysql. In Asterisk1.8 the collum duration and billsec have milisec durations if you include the hrtimers option. On Sun, Feb 13, 2011 at 6:59 PM, Tilghman Lesher tilgh...@meg.abyt.eswrote: On Thursday 10 February 2011 12:33:40 Rodrigo Lang wrote: 2011/2/10 Tilghman Lesher tilgh...@meg.abyt.es On Thursday 10 February 2011 06:13:38 Rodrigo Lang wrote: I wonder if it is possible, without touching the source code, to Asterisk save the cdr with date in unix time instead of the default date. It's possible? The answer is, it depends upon the backend version you're using. With cdr_pgsql and cdr_mysql from 1.6.2 forward, if the column type is integer or float, then the unix timestamp will be used. Without any modification? Only with the column type, Asterisk will modify the common date to unix time? The idea behind this is that we don't want to lose any information. Thus, if the datatype is numeric, then the only way to ensure that we don't lose information during the insert is to set the data to a unixtime format. Note that we can even store fractions of a second in this way, if the column type supports it (i.e. decimal or float). -- Tilghman -- _ -- 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 -- Robert -- _ -- 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 IP335
On 02/17/2011 12:10 AM, Ryan Wagoner wrote: [snip] Backlight works fine on a IP550 with 3.3.1 . I have mine set to off when idle. I like that the 3.3.x series doesn't required the default sip.cfg and phone1.cfg files. The structure of the XML seems cleaner and more consistent. up up.backlight up.backlight.idleIntensity=0 up.backlight.onIntensity=3 /up.backlight /up Here's what I have: up up.idleTimeout=10 up.backlight.idleIntensity=0 up.backlight.onIntensity=3 / That's obviously using a different way (is syntax the proper word?). Don't know if that could make a difference. The config does work except for this setting and the MWI chirp. The only bug I've seen with 3.3.1 is on the IP335. After dialing when it connects the caller name and number jump 1 pixel higher, which looks weird as it is close to the line. One 3.2.3 it didn't move up and looked centered. However the scrolling caller id for incoming calls make this minor annoyance worth the upgrade. That's good to know. Thanks for the info. Regards, Patrick -- _ -- 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] Google 10%
Any thoughts? http://googleblog.blogspot.com/2011/02/simple-way-for-publishers-to-mana ge.html 10% sounds like a bargain for what amounts to a license server Am i missing something? I'm wondering how this can be used on the asterisk platform? Cheers, Dean Posted By Dean Collins to Dean Collins http://blog.collins.net.pr/2011/02/google-10.html at 2/16/2011 08:43:00 PM -- _ -- 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 IP335
On 02/17/2011 12:17 AM, ERIC HERRON wrote: I haven’t played with the backlights yet. One annoyance at a time. Agreed :) To disabled the mwi chirp can be set to silence. MESSAGE_WAITING se.pat.misc.1.name=message waiting se.pat.misc.1.inst.1.type=silence …. Thanks for the tip Eric. The MESSAGE_WAITING part was not what I had in my config. I'll give it a try. I am trying the different firmwares now to see if it makes any difference. Here's my IP670 config re MWI which works with Asterisk 1.4.something: msg msg.bypassInstantMessage=1 msg.mwi.1.callBackMode=contact msg.mwi.1.subscribe=extension msg.mwi.1.callBack=VM extension / Pressing Messages will go straight to Asterisk's VM app and when I have voicemail I don't see a Msgs softkey. Have you checked the logfiles (if using e.g. tftp)? If anything was wrong with the config the logfiles should tell you. Hope you get it working. Please keep us updated on the list. If you hit a brick wall I'll fire up my IP335 and have a look too. Imho judging from user feedback anything moving or scrolling on a phone screen is annoying, distracts and should be disabled. I find the blinking MWI led annoying and afaict there's no way to change its behavior. Regards, Patrick -- _ -- 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 IP335
On 02/17/2011 12:17 AM, ERIC HERRON wrote: [snip] I am trying the different firmwares now to see if it makes any difference. In the admin guide I just came across: up.oneTouchVoiceMail default 0 If set to 1, the voice mail summary display is bypassed and voice mail is dialed directly (if configured). Hope this helps. Regards, Patrick -- _ -- 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 IP335
On Wed, Feb 16, 2011 at 8:38 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 02/17/2011 12:10 AM, Ryan Wagoner wrote: up up.backlight up.backlight.idleIntensity=0 up.backlight.onIntensity=3 /up.backlight /up Here's what I have: up up.idleTimeout=10 up.backlight.idleIntensity=0 up.backlight.onIntensity=3 / That's obviously using a different way (is syntax the proper word?). Don't know if that could make a difference. The config does work except for this setting and the MWI chirp. Your config looks fine to me. For the 3.3.x series they changed how the xml was grouped. For a setting like x.y.z it used to just be x x.y.z=value / now it is xx.y x.y.z=value/x.y/x. From what I have noticed the phone only cares about the x.y.z=value and not which section it is under. My 3.2.x config file worked except for alert info, ringer, and feature settings, which was outlined in Simplified_Configuration_Improvements_in_UC_Software3_3_0_TB60519.pdf Either way check the log the phone uploads on the provisioning server. It will tell you which parameters were rejected. You can also find the number of parameters accepted in rejected in the phone's menu. Ryan -- _ -- 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 IP335
If set to 0, when you press Msgs, it goes to the message center. If set to 1, it dials the voicemail box directly bypassing the message center. I been all over..lol. Thanks though! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Lists Sent: Wednesday, February 16, 2011 9:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom IP335 On 02/17/2011 12:17 AM, ERIC HERRON wrote: [snip] I am trying the different firmwares now to see if it makes any difference. In the admin guide I just came across: up.oneTouchVoiceMail default 0 If set to 1, the voice mail summary display is bypassed and voice mail is dialed directly (if configured). Hope this helps. Regards, Patrick -- _ -- 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 _ No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1204 / Virus Database: 1435/3448 - Release Date: 02/16/11 -- _ -- 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] Hide the plain text password
On Tue, Feb 15, 2011 at 10:31 AM, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Tuesday, February 15, 2011 9:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hide the plain text password Security through obscurity does not work with open source software. What a bold statement, are you telling me it works with closed source software? :P I love this, here you go, security through obscurity at its best: http://www.feplaw.com/news/lawsuit-filed-against-kaba-ilco20110211.cfm -- _ -- 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 IP335
On 02/17/2011 03:20 AM, Ryan Wagoner wrote: [snip] whichsection it is under. My 3.2.x config file worked except for alert info, ringer, and feature settings, which was outlined in Simplified_Configuration_Improvements_in_UC_Software3_3_0_TB60519.pdf Just went through that doc. Interesting read. When I came across ind.pattern I had some hope that the MWI LED could be changed. Unfortunately it seems ind.pattern is missing in 3.3.1 firmware. Either way check the log the phone uploads on the provisioning server. It will tell you which parameters were rejected. You can also find the number of parameters accepted in rejected in the phone's menu. No parameters were rejected. Maybe my perception of backlight off is incorrect. When it is off I expect it so be similar to a Cisco 7961. So no light whatsoever and very hard to read in dim light. Yet in the Idle state the screen of the IP670, to me, still looks like it is still lit and I can clearly read anything that's on the screen. Made pics of backlight off in idle state and on. Am I missing something? http://www.xs4all.nl/~pjl/tmp/IP670_backlight_off.jpg http://www.xs4all.nl/~pjl/tmp/IP670_backlight_on.jpg Regards. Patrick -- _ -- 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 IP335
On 02/17/2011 12:17 AM, ERIC HERRON wrote: To disabled the mwi chirp can be set to silence. MESSAGE_WAITING se.pat.misc.1.name=message waiting se.pat.misc.1.inst.1.type=silence …. This did not work but looking at the example files in the 3.3.1 firmware the snippet below did work (mind the line wrap!). Well, it no longer makes the stupid sound but it still gets called so it wakes up the phone from the idle state so the screen's backlight is turned on and stays that way until the idle timer expires and backlight turns off again. There really should be an option to turn this off without disturbing idle state. se se.pat se.pat.misc se.pat.misc.messageWaiting se.pat.misc.messageWaiting.name=message waiting se.pat.misc.messageWaiting.inst se.pat.misc.messageWaiting.inst.1.atten=0 se.pat.misc.messageWaiting.inst.1.param=0 se.pat.misc.messageWaiting.inst.1.type=silence se.pat.misc.messageWaiting.inst.1.value=0 se.pat.misc.messageWaiting.inst.2.atten=0 se.pat.misc.messageWaiting.inst.2.param=0 se.pat.misc.messageWaiting.inst.2.type=silence se.pat.misc.messageWaiting.inst.2.value=0 se.pat.misc.messageWaiting.inst.3.type=silence se.pat.misc.messageWaiting.inst.3.value=0 /se.pat.misc.messageWaiting.inst /se.pat.misc.messageWaiting /se.pat.misc /se.pat /se Regards, Patrick -- _ -- 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] Asterisk Using as a SIP Client
Hi I wanted to use asterisk as SIP client in my centOS box.I should able to make calls and receive calls.and should able to talk and listen from the headset that I connected to my CentOS box. I need a direction to start on this. Thanks Nikhil -- _ -- 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 Using as a SIP Client
Install asterisknow and begin from there. http://www.asterisk.org/asterisknow/ and don’t miss to read the documentation https://wiki.asterisk.org/wiki/display/AST/Home Regards Khaled Chehab NGN Eng. Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nikhil Sent: Thursday, February 17, 2011 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Using as a SIP Client Hi I wanted to use asterisk as SIP client in my centOS box.I should able to make calls and receive calls.and should able to talk and listen from the headset that I connected to my CentOS box. I need a direction to start on this. Thanks Nikhil -- _ -- 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 * 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 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
