Re: [asterisk-users] switch QOS requirements
Al lists [EMAIL PROTECTED] writes: Its much more reliable than translating DSCP to COS by switch which i'm not sure which switch does that and which one doesn't COS only works if you use a tagged interface on your Asterisk machine. Untagged packets have nowhere to put the COS tag. It also doesn't survive routing (obviously, since it's layer 2). /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] External MWI question for Asterisk
Jason Crum [EMAIL PROTECTED] writes: Gah. So currently in 1.4, there is no method of having Asterisk accept SIP NOTIFY from another server, and pass it on to endpoints if it matches? I can't imagine this being that complex, but then again I'm not familiar with the Asterisk internals. It just seems Asterisk would compare the SIP NOTIFY to what it has currently registered (sip show peers) and forward it on to the endpoint. I'm pretty sure sipXecs can do this. Anyway, thanks for the reply Olle. I think if I re-design my solution for the phones to register with sipXecs and not Asterisk I might make some headway, so that's my next move. What do you gain from having Asterisk at all, if you use sipXecs? /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to register h323 users?
Hi, I have to register h323 users in Asterisk. Please help me in finding out which configuration file to configure. Will it work with gnugk? I am using SJphone as softphone. I am calling users after registering there in the phone and using the ip address of the other system. But I did not register any user in h323.conf or ooh323.conf. Please help me regarding this. Thanking you, With regards, Preeta Pandey Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_valetparking.c anyone using it on 1.4?
ValetParking doesn't announce anything because the whole point of ValetParking is to be able to explicitly park a call at a known spot. I was under the impression that the Valet part of ValetParking meant that you *don't* explicitly park a call at a known spot - the valet takes your call, finds a free spot for it and then tells you where it has been parked. Of course, parking voice may be nothing like parking vehicles :) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of marvin horst Sent: 05 February 2008 13:32 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_valetparking.c anyone using it on 1.4? Hi List, I have this running, but after I park a call it will not announce where it is at, it's like you have to call another application just to say where it is parked at. I have tried a second priority option for the same extension with that ValetParkList but it seems once ValetParkCall has been ended it will not process anymore priorities in this extension. Any ideals or help would be great! I'm using ValetParking with 1.4. ValetParking doesn't announce anything because the whole point of ValetParking is to be able to explicitly park a call at a known spot. We use it to park a call at a users extension when they aren't at their desk, or are on another call. When they're finished with the call or they're paged all they have to do is dial their known park location. I haven't had any problems with priority options. -- Marvin Horst ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_valetparking.c anyone using it on 1.4?
On Feb 6, 2008 3:46 AM, Steve Langstaff [EMAIL PROTECTED] wrote: ValetParking doesn't announce anything because the whole point of ValetParking is to be able to explicitly park a call at a known spot. I was under the impression that the Valet part of ValetParking meant that you *don't* explicitly park a call at a known spot - the valet takes your call, finds a free spot for it and then tells you where it has been parked. Is that what the builtin function ParkAndAnnounce does? Of course, parking voice may be nothing like parking vehicles :) SInce when has anything dealing with computer systems made sense? :-) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of marvin horst Sent: 05 February 2008 13:32 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_valetparking.c anyone using it on 1.4? Hi List, I have this running, but after I park a call it will not announce where it is at, it's like you have to call another application just to say where it is parked at. I have tried a second priority option for the same extension with that ValetParkList but it seems once ValetParkCall has been ended it will not process anymore priorities in this extension. Any ideals or help would be great! I'm using ValetParking with 1.4. ValetParking doesn't announce anything because the whole point of ValetParking is to be able to explicitly park a call at a known spot. We use it to park a call at a users extension when they aren't at their desk, or are on another call. When they're finished with the call or they're paged all they have to do is dial their known park location. I haven't had any problems with priority options. -- Marvin Horst ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Somewhere I wish I wasn't ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Directing SIP/RTP sessions b/w UA
Hi, Let me explain what I'm looking for a solution using asterisk. I have one third party SIP based server (A) and on Asterisk server (B). 1. Extension-1 -- Server A calls Server B. 2. Server B does some processing and calls/sends back to Server A --- Extension-2 3. SIP session has been established b/w two Extension-1 and Extension-2. Now is there any config that I can do in sip.conf which causes direct sip/rtp communication between Extension-1 and Extension-2 without involving Server-B Exten-1--- | | Server A | |ServerB | Exten-2--- | -ag ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wireless VOIP phone recommendations?
Is the new Gigaset S675 IP actually available? And has anyone tried it? From what I've heard, yes in Germany. I'm waiting for our preferred supplier to get some over here in the UK, then I can give it a test. I can't find it available in the US. I'm wondering if it's worth waiting or should I just get one of the older models? From what I've been told it's probably still a few weeks away, so if you have an urgent requirement, I'd go with the S450 for now. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Softphones] ZoIPer vs. XLite?
On Tue, 5 Feb 2008 13:56:37 + (GMT), Tim H. Panton [EMAIL PROTECTED] wrote: Jared was talking about a decent IAX hardphone on this list a week or so back, I don't recall the make. Google didn't return anything with Jared IAX in the gmane.comp.telephony.pbx.asterisk.user archives. You should not need to make _any_ changes to the firewall at the remote end (unless they block all outgoing UDP). Thanks. BTW, will Asterisk 1.6 support STUN so that the server can punch out UDP port for RTP like SIP clients do? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_valetparking.c anyone using it on 1.4?
-Original Message- From: Lacy Moore Sent: 06 February 2008 10:54 On Feb 6, 2008 3:46 AM, Steve Langstaff [EMAIL PROTECTED] wrote: ValetParking doesn't announce anything because the whole point of ValetParking is to be able to explicitly park a call at a known spot. I was under the impression that the Valet part of ValetParking meant that you *don't* explicitly park a call at a known spot - the valet takes your call, finds a free spot for it and then tells you where it has been parked. Is that what the builtin function ParkAndAnnounce does? Of course, parking voice may be nothing like parking vehicles :) SInce when has anything dealing with computer systems made sense? :-) Oh, I feel like such a fool! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem forwarding a call with an AGI script
Hi,
I'm trying to achieve the following:
Incoming call for user A (97), user A make a blind transfer to user B's
phone (96).
User B's phone rings and since there is no one to take the call, it
returns the call to User A with an AGI script.
The dialplan looks like this:
[local]
exten = 96,1,Dial(SIP/user4,10,tr)
exten = 96,2,AGI(transfer.php)
exten = 97,1,NoOp(MARKE1)
exten = 97,2,DIAL(SIP/user1,20,tr)
exten = 97,3,NoOp(MARKE2-)
exten = 97,4,BUSY()
transfer.php:
#! /usr/bin/php -q
?php
$i=0;
ob_implicit_flush(true);
set_time_limit(6);
error_reporting(0);
$stdin=fopen(php://stdin,r);
while(!feof($stdin)){
$temp = fgets($stdin);
$temp = str_replace(\n,,$temp);
$s = explode(:,$temp);
if( $s[0]==agi_dnid){
// $s[1] contains the number that has forwarded the call
$stdout=fopen(php://stdout,w);
fwrite($stdout,NOOP(dummy));
fwrite($stdout,SET CONTEXT local);
fwrite($stdout,SET EXTENSION $s[1]);
fwrite($stdout,SET PRIORITY 1);
fclose($stdout);
}
}
fclose($stdin);
?
And here is the output on the cli:
-- Launched AGI Script /var/lib/asterisk/agi-bin/transfer.php
AGI Tx agi_request: transfer.php
AGI Tx agi_channel: Local/[EMAIL PROTECTED],2
AGI Tx agi_language: en
AGI Tx agi_type: Local
AGI Tx agi_uniqueid: asterisk-1202301352.342
AGI Tx agi_callerid: 98
AGI Tx agi_calleridname: Stefan Guenther
AGI Tx agi_callingpres: 0
AGI Tx agi_callingani2: 0
AGI Tx agi_callington: 0
AGI Tx agi_callingtns: 0
AGI Tx agi_dnid: unknown
AGI Tx agi_rdnis: 97
AGI Tx agi_context: local
AGI Tx agi_extension: 96
AGI Tx agi_priority: 2
AGI Tx agi_enhanced: 0.0
AGI Tx agi_accountcode:
AGI Tx
AGI Rx LI
AGI Tx 510 Invalid or unknown command
-- Nobody picked up in 2 ms
-- Executing [EMAIL PROTECTED]:3] NoOp(SIP/sguenther-08251bf0,
MARKE2-) in new stack
-- Executing [EMAIL PROTECTED]:4] Answer(SIP/sguenther-08251bf0, ) in
new stack
== Auto fallthrough, channel 'SIP/sguenther-08251bf0' status is
'NOANSWER'
-- Executing [EMAIL PROTECTED]:1] DeadAGI(SIP/sguenther-08251bf0,
hangup.php) in new stack
Why doesn't the script jump to priority 1 in extension 97?
Is there really an Invalid or unknown command, I couldn't find one?
Thanks for your help,
Stefan
--
in-put GbR - Das Linux-Systemhaus
Stefan-Michael Guenther
Geschaeftsfuehrer
Moltkestrasse 49 D-76133 Karlsruhe
Tel./Fax : +49 (0)721 / 83044 - 98/93
http://www.in-put.de
Schulungen Installationen
Beratung Support
Voice-over-IP-Loesungen
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R2 with Alestra in Mexico...
Please, Give us more information about error. Are you using astunicall ? 2008/2/5, Carlos Chavez [EMAIL PROTECTED]: I am trying to set up Astunicall 1.4.16 with a link from Alestra in Mexico City. I have done everything I usually do for other links in Mexico but this one simply will not send or receive calls. I just get Protocol error. Anyone has any experience with R2 and Alestra? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R2 with Alestra in Mexico...
Carlos, I have some spare time today in case you want me to check it. Is this your first time with Alestra? On Feb 5, 2008 6:50 PM, Carlos Chavez [EMAIL PROTECTED] wrote: I am trying to set up Astunicall 1.4.16 with a link from Alestra in Mexico City. I have done everything I usually do for other links in Mexico but this one simply will not send or receive calls. I just get Protocol error. Anyone has any experience with R2 and Alestra? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] External MWI question for Asterisk
5 feb 2008 kl. 20.13 skrev Jason Crum: Gah. So currently in 1.4, there is no method of having Asterisk accept SIP NOTIFY from another server, and pass it on to endpoints if it matches? I can't imagine this being that complex, but then again I'm not familiar with the Asterisk internals. It just seems Asterisk would compare the SIP NOTIFY to what it has currently registered (sip show peers) and forward it on to the endpoint. I'm pretty sure sipXecs can do this. But you have to consider that you are mixing two namespaces in Asterisk. SIPx is a SIP-only proxy/PBX, but Asterisk is multiprotocol. We have the dialplan, which is the address you call. And then device names, which are more accounts. Those are separate. A notify is addressed to a SIP uri, something that we handle as an extension. We need to route it internally. You might want to send a SIP MWI notify to a hotel phone connected to a channel bank to get those irritating red lamps to blink... You can't think SIP only in Asterisk (even though I sometimes like the idea) since we've got a multiprotocol architecture. /O Anyway, thanks for the reply Olle. I think if I re-design my solution for the phones to register with sipXecs and not Asterisk I might make some headway, so that's my next move. On Feb 5, 2008 1:52 AM, Johansson Olle E [EMAIL PROTECTED] wrote: It is currently not possible. With the new event-driven MWI notification system in 1.6, it should be possible to add code for it, but it would be kind of tricky. If you send an MWI to an extension - how do we know where to send it? We either need to use the existing hints that connect the extensions to the device name space, or add a new sort of voicemail hints that connects an extension to a voicemailbox ID that we devices can subscribe to. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- * Olle E Johansson - [EMAIL PROTECTED] * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT] ISDN 30 (PRI) service in the Netherlands
Hello, and please forgive the OT question. I'm just becoming desperate. I need two ISDN 30 circuits in Amsterdam, and I can't seem to be able to get a provider. I've tried KPN and Versatel. I'm based in California. Does anyone have any recommendations? Thanks in advance, Jose. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R2 with Alestra in Mexico...
On Wed, 2008-02-06 at 08:17 -0600, Moises Silva wrote: Carlos, I have some spare time today in case you want me to check it. Is this your first time with Alestra? Thank you for the offer. Yes this is the first time I use Alestra for R2. I have another customer that uses them but with PRI and I do have some problems dialing certain numbers on that link. It turns out that there was a problem with their equipment but it took them almost 24 hours for them to admit it. It is now working properly. Calls now go in and out and for now I do not see any other problems. My list of tested providers for R2 in Mexico is now: Axtel, Alestra, Maxcom and Telmex. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] ISDN 30 (PRI) service in the Netherlands
On 09:00, Wed 06 Feb 08, Jose Quinteiro wrote: Hello, and please forgive the OT question. I'm just becoming desperate. I need two ISDN 30 circuits in Amsterdam, and I can't seem to be able to get a provider. I've tried KPN and Versatel. I'm based in California. Does anyone have any recommendations? You can only get those lines here in .nl if you have an address here. So you need a dutch company or a company with a dutch location. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R2 with Alestra in Mexico...
This is great news :) On Feb 6, 2008 10:56 AM, Carlos Chavez [EMAIL PROTECTED] wrote: On Wed, 2008-02-06 at 08:17 -0600, Moises Silva wrote: Carlos, I have some spare time today in case you want me to check it. Is this your first time with Alestra? Thank you for the offer. Yes this is the first time I use Alestra for R2. I have another customer that uses them but with PRI and I do have some problems dialing certain numbers on that link. It turns out that there was a problem with their equipment but it took them almost 24 hours for them to admit it. It is now working properly. Calls now go in and out and for now I do not see any other problems. My list of tested providers for R2 in Mexico is now: Axtel, Alestra, Maxcom and Telmex. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P phone won't ring
Shane Wegner wrote: Hello all, I have two handsets connected to FXS ports on a TDM400P, both GE models but one rings and the other does not. The phone models are not identical. The phone which doesn't ring on the TDM does ring when connected to a regular POTS line and I tried connecting another phone to the port and it rings fine. Do you have the power connector on the TDM400P card hooked up? Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] ISDN 30 (PRI) service in the Netherlands
We have space at a co-location facility there (Telecity). That's not good enough? Michiel van Baak wrote: On 09:00, Wed 06 Feb 08, Jose Quinteiro wrote: Hello, and please forgive the OT question. I'm just becoming desperate. I need two ISDN 30 circuits in Amsterdam, and I can't seem to be able to get a provider. I've tried KPN and Versatel. I'm based in California. Does anyone have any recommendations? You can only get those lines here in .nl if you have an address here. So you need a dutch company or a company with a dutch location. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] ISDN 30 (PRI) service in the Netherlands
On 10:27, Wed 06 Feb 08, Jose Quinteiro wrote: We have space at a co-location facility there (Telecity). That's not good enough? Dont know for sure, but I'm afraid not. Cant the ppl at Telecity help you out on this issue ? Michiel van Baak wrote: On 09:00, Wed 06 Feb 08, Jose Quinteiro wrote: Hello, and please forgive the OT question. I'm just becoming desperate. I need two ISDN 30 circuits in Amsterdam, and I can't seem to be able to get a provider. I've tried KPN and Versatel. I'm based in California. Does anyone have any recommendations? You can only get those lines here in .nl if you have an address here. So you need a dutch company or a company with a dutch location. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to hookup to cell phone for outbound calls?
On Feb 5, 2008 9:10 PM, Ed W [EMAIL PROTECTED] wrote: I need a small PBX for use on the move. This means that outbound calls will need to be made over the cell phone network. What's your budget? You could use voiceblue's SIP/GSM gateway (exists in 2 or 4 channels), it connects to your internal NIC of your small pbx (like in: laptop?) or by using a switch.. cheers, stoffell ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom BLF / Speed Dial
Is there a way to configure the buttons on the phone that are normally reserved for line registrations so that I can do a one-button pickup of a parked call complete with Presence? The goal is to have a couple of the line registration buttons show me who is on park orbits 701 and 702 so that I can pick them up with one-touch. Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] Attachment encrypted? click here. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] ISDN 30 (PRI) service in the Netherlands
Jose Quinteiro wrote: We have space at a co-location facility there (Telecity). That's not good enough? No, that's not good enough. Please contact me off-list. We might be able to help you. Ron Arts NeoNova Michiel van Baak wrote: On 09:00, Wed 06 Feb 08, Jose Quinteiro wrote: Hello, and please forgive the OT question. I'm just becoming desperate. I need two ISDN 30 circuits in Amsterdam, and I can't seem to be able to get a provider. I've tried KPN and Versatel. I'm based in California. Does anyone have any recommendations? You can only get those lines here in .nl if you have an address here. So you need a dutch company or a company with a dutch location. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cannot hear voice through SIP Phone from one side
On Feb 5, 2008 9:52 PM, Sanjoy Rath [EMAIL PROTECTED] wrote: Any suggestion how to get it to work :) Look at the transmit silence option in X-Lite and make sure that the client *does* transmit silence, not attempt to keep quiet. I can't remember if checking the box transmits or does not transmit: you want it to transmit silence. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Softphones] ZoIPer vs. XLite?
Vincent, the phone referred to that Jared mentioned is the Allnet 7960. I have an ongoing review of it here (meaning I never finished it properly). http://food4wine.ning.com/ On Feb 6, 2008 12:44 PM, Vincent [EMAIL PROTECTED] wrote: On Tue, 5 Feb 2008 13:56:37 + (GMT), Tim H. Panton [EMAIL PROTECTED] wrote: Jared was talking about a decent IAX hardphone on this list a week or so back, I don't recall the make. Google didn't return anything with Jared IAX in the gmane.comp.telephony.pbx.asterisk.user archives. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need a dial rule to match and replace a number.
I am using Asterisk 1.2.18 with FreePBX 2.2.0. I have two Asterisk systems with an IAX2 trunk between them. I want to make each end so when a user dials the local 7 digit number for the other end it will try to rute the call through the IAX2 trunk before trying the PSTN lines. When the call comes in on the other end I want it to hit my external IVR. The IAX2 trunk connection is working great a call going to 1234567 goes over the Internet to the other end but then on the receiving end it tries to dial out a zap channel to call back in the 1234567 zap channel. I have the outbound route to match 1234567 to the IAX2 trunk and in the IAX2 trunk I need to strip off the 7 digit number and replace it with a *02 to call my external IVR on the other end. For testing I have been trying to make the call connect to my extension on the other end but I am not having any luck. I need some help to make this Dial Rule work. This is what I am trying to use 1234567|+219 From what I have read this should strip off the exact matching 7 digit local number 1234567 and add a prefix of 219 but it does not. From trying different orders and mixing this around I am only able to do one or the other. I can either strip off the 1234567 which does nothing or I can add a 219 prefix that calls 2191234567 on the other end. What do I need to do to strip off the 7 digit number that was dialed and replace it with a 219 or replace it with a *02? -- Open Source: To innovate then create Proprietary: To imitate then litigate ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FXO modules and polarity reverse
I would like to know if any body tried to connect gsm gateway with polarity reversal to fxo module at asterisk server ,and if the polarity reversal solve the problem of the answer and hangup supervison on calls .i appreciate any help.Thanks in advance;Wassim _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P phone won't ring
Have you swapped the phones between the FXS ports to see if the phone rings? Moj Shane Wegner wrote: Hello all, I have two handsets connected to FXS ports on a TDM400P, both GE models but one rings and the other does not. The phone models are not identical. The phone which doesn't ring on the TDM does ring when connected to a regular POTS line and I tried connecting another phone to the port and it rings fine. So, I'm presuming the TDM is ringing the handsets somehow differently than the telco in a way which most phones like but this particular one doesn't deal with. On the wctdm module I've tried ringboost=1 and fastringer=1 but neither made a difference, not that fastringer=1 should as I'm in Canada where we use 20HZ I believe. Just wondering if there are any other settings in the Zaptel modules or Asterisk to change the ring properties. Best, Shane ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO modules and polarity reverse
On Wed, Feb 06, 2008 at 09:54:14PM +0200, wassim darwish wrote: I would like to know if any body tried to connect gsm gateway with polarity reversal to fxo module at asterisk server ,and if the polarity reversal solve the problem of the answer and hangup supervison on calls .i appreciate any help.Thanks in advance;Wassim Which FXO adapter? I hope you realise that the X100P that our Impartial Sam is pushing for to connect to some GSM gateways does not support detecting polarity reversal. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI Process Count (HOWTO?)
Is there any way to see the number of AGI processes that Asterisk is handling? Either console, Asterisk Manager, or from within the AGI? I used to just count the number of running copies of my AGI process (ps aux | grep agi) but once in a blue moon one of my AGI processes will become a zombie or for some other reason not stop when Asterisk disconnects from it. I'd like to know, from Asterisk's point of view, the number of external applications it's communicating with. -- /Nick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gemeinschaft released
Hello, Do I need any Asterisk Patches? I had a look to the source code, specially the provisioning system. Not very readable, no classes and many code lines commented. The cluster capability is very interesting. Regards, Gunnar Schaller Hi, Just wanted to let you know that we have just made our GPL toolkit Gemeinschaft available to the public. (Finally.) Mostly German for now - about half of the strings in the language strings file have been translated to English. I'm a software developer, not a marketing guy, so ... svn co https://svn.amooma.de/gemeinschaft/trunk gemeinschaft-trunk German readers: see http://www.amooma.de/gemeinschaft/ Regards, Philipp Kempgen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE412P and Delll PowerEdge 2900
Hello, Looking for comments if Digium TE412P (32-bit 33MHz card keyed for 3.3 volt operation) compatible with Dell PowerEdge 2900 server board (1 PCI Express X8, 3 PCI Express X4, 2 64-bit/133MHz PCI-X)? Any know issue with Digium cards for this server family? Thanks in advance. Ash. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] External MWI question for Asterisk
My first post on this subject describes the environment in which I'm using sipX. I'm only using sipX as a method of getting UDP to TCP and back again. This is only a test lab I'm working with, but in production we use Asterisk and several thousand dollars worth of Digium equipment to connect to the PRI and echo cancellation hardware, so there's that ;) On Feb 6, 2008 4:09 AM, Benny Amorsen [EMAIL PROTECTED] wrote: What do you gain from having Asterisk at all, if you use sipXecs? /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] External MWI question for Asterisk
Very good point Olle. Guess I'm overlooking the other potential MWI signaling aside from SIP. Thanks for your input! On Feb 6, 2008 10:22 AM, Johansson Olle E [EMAIL PROTECTED] wrote: But you have to consider that you are mixing two namespaces in Asterisk. SIPx is a SIP-only proxy/PBX, but Asterisk is multiprotocol. We have the dialplan, which is the address you call. And then device names, which are more accounts. Those are separate. A notify is addressed to a SIP uri, something that we handle as an extension. We need to route it internally. You might want to send a SIP MWI notify to a hotel phone connected to a channel bank to get those irritating red lamps to blink... You can't think SIP only in Asterisk (even though I sometimes like the idea) since we've got a multiprotocol architecture. /O Anyway, thanks for the reply Olle. I think if I re-design my solution for the phones to register with sipXecs and not Asterisk I might make some headway, so that's my next move. On Feb 5, 2008 1:52 AM, Johansson Olle E [EMAIL PROTECTED] wrote: It is currently not possible. With the new event-driven MWI notification system in 1.6, it should be possible to add code for it, but it would be kind of tricky. If you send an MWI to an extension - how do we know where to send it? We either need to use the existing hints that connect the extensions to the device name space, or add a new sort of voicemail hints that connects an extension to a voicemailbox ID that we devices can subscribe to. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- * Olle E Johansson - [EMAIL PROTECTED] * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Real API for Perl?
Alex Balashov wrote: Well, no, there really aren't any prebuilt high-level frameworks for approaching Asterisk through the Manager API or AGI. There is actually a couple of CPAN packages for interacting with the AMI in an event oriented fashion. http://search.cpan.org/search?query=asteriskmode=all Enjoy! -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gemeinschaft released
Gunnar Schaller wrote: Do I need any Asterisk Patches? No. Just a vanilla Asterisk 1.4. I had a look to the source code, specially the provisioning system. Not very readable, no classes and many code lines commented. Well, we're still working on it. Classes don't _necessarily_ make code better, it's just hip to use OO. Patches are welcome :) The cluster capability is very interesting. Exactly. Although I must admit wo don't really have much documentation on how to install a Gemeinschaft cluster. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI Process Count (HOWTO?)
On Wednesday 06 February 2008 15:09:06 Nicholas Blasgen wrote: Is there any way to see the number of AGI processes that Asterisk is handling? Either console, Asterisk Manager, or from within the AGI? I used to just count the number of running copies of my AGI process (ps aux | grep agi) but once in a blue moon one of my AGI processes will become a zombie or for some other reason not stop when Asterisk disconnects from it. I'd like to know, from Asterisk's point of view, the number of external applications it's communicating with. If you type 'show channels' and count the number of invocations of AGI in the output, you'll have your count. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Post Call QoS....?
Ok, so I've asked this question before, and didn't get an answer. So here I go again! Asterisk 1.4 has some channel variables that you can inspect after a call is complete that will give you QoS metrics. Stuff like average round trip time, etc. Since there's only one set of variables, and calls will have two channels, which channel is this information for? Is it for one of the channels? Is it an aggregate of both channels? Who added this code and what where they thinking when they wrote it? Thanks, Doug. Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R2 with Alestra in Mexico...
Yes, i have the same problem with att a few months ago, the problem is the acount (abonado) code, att need 2 and the code of unicall send 1, maybe the problem is the same for you, please post the debug unicall code. In this code, you can see the dial number, but if you see, the last digit is 1 On Feb 6, 2008 12:21 PM, Moises Silva [EMAIL PROTECTED] wrote: This is great news :) On Feb 6, 2008 10:56 AM, Carlos Chavez [EMAIL PROTECTED] wrote: On Wed, 2008-02-06 at 08:17 -0600, Moises Silva wrote: Carlos, I have some spare time today in case you want me to check it. Is this your first time with Alestra? Thank you for the offer. Yes this is the first time I use Alestra for R2. I have another customer that uses them but with PRI and I do have some problems dialing certain numbers on that link. It turns out that there was a problem with their equipment but it took them almost 24 hours for them to admit it. It is now working properly. Calls now go in and out and for now I do not see any other problems. My list of tested providers for R2 in Mexico is now: Axtel, Alestra, Maxcom and Telmex. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom BLF / Speed Dial
I figured it out. Thanks anyway! Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] Attachment encrypted? click here. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Munger Sent: Wednesday, February 06, 2008 1:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom BLF / Speed Dial Is there a way to configure the buttons on the phone that are normally reserved for line registrations so that I can do a one-button pickup of a parked call complete with Presence? The goal is to have a couple of the line registration buttons show me who is on park orbits 701 and 702 so that I can pick them up with one-touch. Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] Attachment encrypted? click here. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom BLF / Speed Dial
Could you possibly post what steps you took to make this work so others (including myself :-) ) may benefit? Thank you! Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 - Original Message - From: Michael Munger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 6, 2008 5:05:20 PM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] Polycom BLF / Speed Dial I figured it out. Thanks anyway! Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] Attachment encrypted? click here. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Munger Sent: Wednesday, February 06, 2008 1:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom BLF / Speed Dial Is there a way to configure the buttons on the phone that are normally reserved for line registrations so that I can do a one-button pickup of a parked call complete with Presence? The goal is to have a couple of the line registration buttons show me who is on park orbits 701 and 702 so that I can pick them up with one-touch. Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] Attachment encrypted? click here. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R2 with Alestra in Mexico...
When I dial one extension to the other, I get the call go into a HOLD music instead of rining the other extention. Both extensions are SIP Softphone. Following is the Asterisks CLI commandline log -- Executing [EMAIL PROTECTED]:1] Park(SIP/500-08276430, ) in new stack -- Started music on hold, class 'default', on SIP/500-08276430 == Parked SIP/500-08276430 on [EMAIL PROTECTED] Will timeout back to extension [from-internal] s, 1 in 45 seconds-- Added extension '701' priority 1 to parkedcalls == Spawn extension (from-internal, s, 1) exited KEEPALIVE on 'SIP/500-08276430' Any thoughts how to get the call to successfully dial and get both the extensions to talk? Thanks, SR. _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call go into a HOLD music instead
When I dial one extension to the other, I get the call go into a HOLD music instead of rining the other extention. Both extensions are SIP Softphone. Following is the Asterisks CLI commandline log -- Executing [EMAIL PROTECTED]:1] Park(SIP/500-08276430, ) in new stack -- Started music on hold, class 'default', on SIP/500-08276430 == Parked SIP/500-08276430 on [EMAIL PROTECTED] Will timeout back to extension [from-internal] s, 1 in 45 seconds-- Added extension '701' priority 1 to parkedcalls == Spawn extension (from-internal, s, 1) exited KEEPALIVE on 'SIP/500-08276430' Any thoughts how to get the call to successfully dial and get both the extensions to talk? Thanks, SR. _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call go into a HOLD music instead
It appears that the number you're dialing (701) is not an extension or UA but rather a call parking slot. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 - Original Message - From: Sanjoy Rath [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 6, 2008 5:37:34 PM (GMT-0600) America/Chicago Subject: [asterisk-users] Call go into a HOLD music instead When I dial one extension to the other, I get the call go into a HOLD music instead of rining the other extention. Both extensions are SIP Softphone. Following is the Asterisks CLI commandline log -- Executing [EMAIL PROTECTED]:1] Park(SIP/500-08276430, ) in new stack -- Started music on hold, class 'default', on SIP/500-08276430 == Parked SIP/500-08276430 on [EMAIL PROTECTED] . Will timeout back to extension [from-internal] s, 1 in 45 seconds -- Added extension '701' priority 1 to parkedcalls == Spawn extension (from-internal, s, 1) exited KEEPALIVE on 'SIP/500-08276430' Any thoughts how to get the call to successfully dial and get both the extensions to talk? Thanks, SR. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to hookup to cell phone for outbound calls?
Sam Tam wrote: Well I think you need a GSM Gateway You can find some info on cyber-telecom.net For a cheap option you can try a CT-G1000 or CT-G2000 and then plug it in a X100P or something similar then it would be very economical. Yep, this is the kind of thing I am after, except my hardware PBX has limited connectivity and ideally I want a USB or ethernet hookup to the box..? The scenario is basically a small commercial PBX (small form factor) which can be supplied with IP phones and will talk out via a cell phone channel (or via a satellite phone if that's the only option available, but this is out of scope of this question). So basically I want to figure out some options to hookup a GSM cellphone channel to a small form factor asterisk PBX which has limited expansion options (ethernet, USB and mini-PCI - although prefer to use the later for a wifi card...) I only need a single channel of GSM right now (and a single SIM) Any thoughts? Remember this needs to be production quality and priced sensible for a commodity market Thanks for pointers to hardware Ed W ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Matching + characters in dial plan
Can someone please explain how to match a + character in a dial plan (so that I can swap it for the 00 country escape code). In Europe at least the + is a common shortcut for the international prefix (which is 00 in my country). However, my trunk chokes on the + character and all my speed-dials are setup with a + at the start of them... Trying to fix the phone rather than the addressbook... Thanks Ed W ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to hookup to cell phone for outbound calls?
On Thu, 07 Feb 2008 00:14:07 +, Ed W wrote: Sam Tam wrote: Well I think you need a GSM Gateway You can find some info on cyber-telecom.net For a cheap option you can try a CT-G1000 or CT-G2000 and then plug it in a X100P or something similar then it would be very economical. Yep, this is the kind of thing I am after, except my hardware PBX has limited connectivity and ideally I want a USB or ethernet hookup to the box..? The scenario is basically a small commercial PBX (small form factor) which can be supplied with IP phones and will talk out via a cell phone channel (or via a satellite phone if that's the only option available, but this is out of scope of this question). So basically I want to figure out some options to hookup a GSM cellphone channel to a small form factor asterisk PBX which has limited expansion options (ethernet, USB and mini-PCI - although prefer to use the later for a wifi card...) I only need a single channel of GSM right now (and a single SIM) Any thoughts? Remember this needs to be production quality and priced sensible for a commodity market Thanks for pointers to hardware Then use a Sipura SPA-3000 instead of the X100 card. It's a freestanding FXO/FXS SIP interface. Well proven. Not overly expensive. Hardware based SIPGSM gateways seem to start at around $250 US. See www.voip-info.org for a list. Michael -- Michael Graves mgravesatmstvp.com blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is encrypted iax safe and secure?
On Tuesday 05 February 2008 09:22:29 Cavalera Claudio Luigi wrote: Hello, I'm doing some research concerning iax encryption, I haven't find any clients (softphones or hardphones) which implement so I have not tested it yet. There was also this message on asterisk-security mailing list http://archives.free.net.ph/message/20070507.101933.222987b2.en.html which got no answers and this makes me think that this iax encryption is not much interesting for the community. Anyway, in iax specification there is this statement: Only the data portion of the messages are encoded. Which are the consequences of this, is it true as stated on http://www.voip-info.org/wiki/view/IAX+encryption that The calling/called numbers are still passed in the clear over encrypted IAX, so you are still vulnerable to traffic analysis. ? If it's true how to deal with this? Would you consider media payload encryption enough? Maybe it's better to just forget about iax encryption and consider some more general approach like using openvpn http://www.voip-info.org/wiki/view/IAX_OpenVPN ? This half-encrypted iax encryption doesn't make much sense to me, therefore I think there's probably something I'm missing/misunderstanding. Is it important for you to conceal that a call was made from abc to xyz on thus-and-such a date? Or do you merely need to conceal the content of a call? You can already do traffic analysis and figure out that a call occurred, just not what the endpoints are (even if you encrypted the entire link). The only way to get around that is to continuously send random garbage through the pipe at the same rate and consistency as would occur with a real IAX2 call. And the endpoints are only as specific as the systems on either end choose to make them. If you used some system of src/dst obfuscation, you could conceal even that information, though repeated calls to various destinations could still be paired and correlated. IAX2 encryption is designed to obscure the same information as is obscured when you encrypt a call over the PSTN -- the content is protected, but the existence of such a call is not. Remember that a potential attacker will always choose the weakest link, and will probably attack the audio stream at a different location, if she cannot listen to the IP stream directly (such as a true wiretap on an analog endpoint or breaking into one of the two machines involved in the encryption). The idea is to make the IAX2 link unattractive as a potential target of wiretapping (whereas before it would have been the most obvious choice), thus forcing the attacker to choose a different attack scenario. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Matching + characters in dial plan
I made up some dialplan rules to strip the '+' and replace with the
00...
Something like:
exten = _+XX.,1,Dial(zap/g1/00${EXTEN:1})
PaulH
On Thu, 2008-02-07 at 00:18 +, Ed W wrote:
Can someone please explain how to match a + character in a dial plan (so
that I can swap it for the 00 country escape code).
In Europe at least the + is a common shortcut for the international
prefix (which is 00 in my country). However, my trunk chokes on the +
character and all my speed-dials are setup with a + at the start of
them... Trying to fix the phone rather than the addressbook...
Thanks
Ed W
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Matching + characters in dial plan
Am Donnerstag, den 07.02.2008, 00:18 + schrieb Ed W:
Can someone please explain how to match a + character in a dial plan (so
that I can swap it for the 00 country escape code).
In Europe at least the + is a common shortcut for the international
prefix (which is 00 in my country). However, my trunk chokes on the +
character and all my speed-dials are setup with a + at the start of
them... Trying to fix the phone rather than the addressbook...
You should get away with
exten = _+[1-9].,1,Goto(00${EXTEN:1},1)
If you had any special use for triple-0 numbers (as we do), you should
afaik also be able to use
exten = _+.,1,Goto(00${EXTEN:1},1)
We do not allow +0 numbers though because that would contradict the
meaning of a 000 number in our setup. Generally +AABBBCCC is dialled
as 00AABBBCCC, as international phone call, through our outward phone
provider without them noticing any weird + signs.
BR
Anselm
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Matching + characters in dial plan
On Wednesday 06 February 2008 18:18:12 Ed W wrote:
Can someone please explain how to match a + character in a dial plan (so
that I can swap it for the 00 country escape code).
In Europe at least the + is a common shortcut for the international
prefix (which is 00 in my country). However, my trunk chokes on the +
character and all my speed-dials are setup with a + at the start of
them... Trying to fix the phone rather than the addressbook...
exten = _+.,1,Goto(00${EXTEN:1},1)
--
Tilghman
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call go into a HOLD music instead
I am not dialing ext 701 but 700 from 500 ext. Do not know why its going to 701. Thanks, Sanjoy. Date: Wed, 6 Feb 2008 17:40:47 -0600From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Re: [asterisk-users] Call go into a HOLD music instead It appears that the number you're dialing (701) is not an extension or UA but rather a call parking slot.Tim NelsonSystems/Network SupportRockbochs Inc.(218)727-4332- Original Message -From: Sanjoy Rath [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSent: Wednesday, February 6, 2008 5:37:34 PM (GMT-0600) America/ChicagoSubject: [asterisk-users] Call go into a HOLD music instead When I dial one extension to the other, I get the call go into a HOLD music instead of rining the other extention. Both extensions are SIP Softphone. Following is the Asterisks CLI commandline log -- Executing [EMAIL PROTECTED]:1] Park(SIP/500-08276430, ) in new stack -- Started music on hold, class 'default', on SIP/500-08276430 == Parked SIP/500-08276430 on [EMAIL PROTECTED] Will timeout back to extension [from-internal] s, 1 in 45 seconds-- Added extension '701' priority 1 to parkedcalls == Spawn extension (from-internal, s, 1) exited KEEPALIVE on 'SIP/500-08276430' Any thoughts how to get the call to successfully dial and get both the extensions to talk? Thanks, SR. _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call go into a HOLD music instead
Sanjoy Rath wrote: I am not dialing ext 701 but 700 from 500 ext. Do not know why its going to 701. Check your features.conf, 700 by default is the parking extensions. It will place calls on 701 first, and then 702 if 701 isn't available. This can be changed via the features.conf Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Avaya phone system
Is there a way to have Asterisk talk to a Avaya IP Office phone system? If it's possible, where can I find the instructions? Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call go into a HOLD music instead
Thanks Doug for your email. I will assign another extension so that it does not comflict with 700. Also I see CPU being 100% used. DO not know if I can stop something to increase the CPU idle %. Thanks, Sanjoy. Date: Wed, 6 Feb 2008 20:35:13 -0500 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Call go into a HOLD music instead Sanjoy Rath wrote: I am not dialing ext 701 but 700 from 500 ext. Do not know why its going to 701. Check your features.conf, 700 by default is the parking extensions. It will place calls on 701 first, and then 702 if 701 isn't available. This can be changed via the features.conf Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R2 with Alestra in Mexico...
Two weeks from now I will release a chan_unicall driver that allows to change the calling party category from the dial plan and configuration file. Keep posted at http://www.moythreads.com/astunicall/ Regards, Moisés Silva On Feb 6, 2008 5:07 PM, Jorge Cisneros [EMAIL PROTECTED] wrote: Yes, i have the same problem with att a few months ago, the problem is the acount (abonado) code, att need 2 and the code of unicall send 1, maybe the problem is the same for you, please post the debug unicall code. In this code, you can see the dial number, but if you see, the last digit is 1 On Feb 6, 2008 12:21 PM, Moises Silva [EMAIL PROTECTED] wrote: This is great news :) On Feb 6, 2008 10:56 AM, Carlos Chavez [EMAIL PROTECTED] wrote: On Wed, 2008-02-06 at 08:17 -0600, Moises Silva wrote: Carlos, I have some spare time today in case you want me to check it. Is this your first time with Alestra? Thank you for the offer. Yes this is the first time I use Alestra for R2. I have another customer that uses them but with PRI and I do have some problems dialing certain numbers on that link. It turns out that there was a problem with their equipment but it took them almost 24 hours for them to admit it. It is now working properly. Calls now go in and out and for now I do not see any other problems. My list of tested providers for R2 in Mexico is now: Axtel, Alestra, Maxcom and Telmex. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need good voicemail documentation
Hi list, After wrestling with the voicemail system for a while (Asterisk 1.4.14, Debian etch), I got it to work, but I still have lots of questions, like: * Why can't I delete any voicemail messages? (Response: Message undeleted.) * Why can't I listen to the messages in the Old folder? * Why can't I use the advanced options? (Response: I'm sorry, I did not understand your response.) * How come if I put [EMAIL PROTECTED] in my phone's context of sip.conf, do I get an error? (CLI: ...Remote host can't match request NOTIFY to call...) Unfortunately, none of the books and other documentation I've found on the subject goes into enough detail to provide answers to such questions. So, can anyone recommend some good Asterisk voicemail documentation that goes beyond merely scratching the surface? Cheers, Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call go into a HOLD music instead
What hardware are you running at the moment? Cheers, Dan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sanjoy Rath Sent: Thursday, 7 February 2008 12:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call go into a HOLD music instead Thanks Doug for your email. I will assign another extension so that it does not comflict with 700. Also I see CPU being 100% used. DO not know if I can stop something to increase the CPU idle %. Thanks, Sanjoy. Date: Wed, 6 Feb 2008 20:35:13 -0500 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Call go into a HOLD music instead Sanjoy Rath wrote: I am not dialing ext 701 but 700 from 500 ext. Do not know why its going to 701. Check your features.conf, 700 by default is the parking extensions. It will place calls on 701 first, and then 702 if 701 isn't available. This can be changed via the features.conf Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: POTS telephone like the SPA-942?
My wife really likes the fit and feel of my SPA-942. Anyone know of a POTS telephone with similar rugged construction? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New deployment questions
Hi, I'm relatively new to Asterisk and would like to set up an Asterisk box as a PBX for a large organization and also as a switch for a small community so I have a few questions Which if the various flavors of Asterisk is preferred for setting up a 100-user PBX and which is preferred for setting up a 1000-user PBX? Will Asterisk scale to 1000 PBX users, and what type of hardware would be required to achieve this on the X86 platform? Will using gateways like the Quintum A and D series boxes as opposed to plugging TDM cards directly into the Asterisk box improve the scalability of Asterisk, and will I still have to buy G.729a licenses for Asterisk if it is only used as a switch for SIP gateways and phones? What is a good low cost TDM gateway? What is the best solution for user authentication for a small 100-user setup and also for a large 1000-user setup? Can Asterisk be used as a switch for a 10,000-user network? What hardware, billing and authentication systems do you recommend? If I want to implement SS7 signaling to another telco, what gateway solution can I use with Asterisk? Thanks in advance for answering my numerous questions Femi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: POTS telephone like the SPA-942?
On Wed, Feb 06, 2008 at 10:04:57PM -0500, Frank Tarczynski wrote: My wife really likes the fit and feel of my SPA-942. Anyone know of a POTS telephone with similar rugged construction? WECo 2500? :-) Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom BLF / Speed Dial
check here: http://www.voip-info.org/wiki/view/Asterisk+cmd+ParkAndAnnounce On Feb 6, 2008 4:22 PM, Tim Nelson [EMAIL PROTECTED] wrote: Could you possibly post what steps you took to make this work so others (including myself :-) ) may benefit? Thank you! Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 - Original Message - From: Michael Munger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 6, 2008 5:05:20 PM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] Polycom BLF / Speed Dial I figured it out. Thanks anyway! Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] Attachment encrypted? click here. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Munger Sent: Wednesday, February 06, 2008 1:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom BLF / Speed Dial Is there a way to configure the buttons on the phone that are normally reserved for line registrations so that I can do a one-button pickup of a parked call complete with Presence? The goal is to have a couple of the line registration buttons show me who is on park orbits 701 and 702 so that I can pick them up with one-touch. Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] Attachment encrypted? click here. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
