[asterisk-users] R: Asterisk V/s FreeSwitch

From the second link Gilles suggested: Aterisk vs FreeSWITCH [...] The mailing list (FreeSWITCH) is also a very nice place. In both places (IRC and mailing list) they are very friendly and supportive, unlike the Asterisk/Digium community. [...] BAH... -Messaggio originale- Da:

[asterisk-users] SIP Realtime Templates (!)

Hello, Is it possible to assign templates defined in sip.conf to sip realtime peers? There was another mail in 2008 which asked the same question but never received a response. Thanks, Alex -- _ -- Bandwidth and Colocation

[asterisk-users] R: Stun Server

In debian do 'apt-get install stun'. Change /etc/default/stun. Done Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Bryant Zimmerman Inviato: mercoledì 27 luglio 2011 15:30 A: asterisk-users@lists.digium.com Oggetto: Re: [asterisk-users]

[asterisk-users] Browser based SIP UA

Hello, can anyone recommend a browser based SIP client that works well with Asterisk? I need something that requires authentication (based on Asterisks peer name and pass). Thanks in advance! Alex -- _ -- Bandwidth and

[asterisk-users] R: Browser based SIP UA

@lists.digium.com Oggetto: Re: [asterisk-users] Browser based SIP UA On 07/26/2011 10:13 AM, Alexandru Oniciuc wrote: can anyone recommend a browser based SIP client that works well with Asterisk? I need something that requires authentication (based on Asterisks peer name and pass). What do you

[asterisk-users] R: How to remove asterisk ?

What do you mean? Did you installed from sources or distro packet? sources: make uninstall distro: Every distro has its own commands (yum, apt-get ecc) Alex Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di virendra bhati Inviato:

[asterisk-users] R: Nat=yes

Thank you all for your answers, I will stick with nat=yes. Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

[asterisk-users] R: R: No Internet, no asterisk

- Non-Commercial Discussion Oggetto: Re: [asterisk-users] R: No Internet, no asterisk Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto: Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and internet is offline. srvlookup = no didn't help. What about putting my provider's

[asterisk-users] R: No Internet, no asterisk

Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and internet is offline. Regards, Alex -Messaggio originale- Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Niccolò Belli Inviato: lunedì 18 aprile 2011

[asterisk-users] SIP Registrations

Hello list, I need some light regarding the way asterisk is handling the SIP Registration method: I have an asterisk 1.6.0.22 and a UAC that sends REGISTER requests without the Authentication part in the sip message. The UAC expects a 401 reply to create the

[asterisk-users] R: asterisk compatible cards?

Hi Daniel, have a look at this page, maybe it will help you find a reseller: http://www.voip-info.org/wiki/view/Asterisk+Consultants+Romania . Best Regards, Alex -Messaggio originale- Da: asterisk-users-boun...@lists.digium.com

[asterisk-users] Group call limit

Hello list, is it possible to group some peers and limit their overall call limit? Ex: 4 peers can make max 2 concurrent calls. Thanks in advance, Alex -- _ -- Bandwidth and Colocation

[asterisk-users] Little t38 bug?

Hello List, I think I've discovered a little bug in t.38 bug in 1.6.0.22 regarding the speed (T38MaxBitRate) used to send the faxes. Asterisk always responds with a=T38MaxBitRate:2400. I've tried with Patton and Grandstream devices and the result is always the

[asterisk-users] R: new way of asterisk and kamailio (openser) realtime integration

kb.asipto.com isn't reachable: DNS doesn't resolve the domain name. Alex -Messaggio originale- Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Daniel-Constantin Mierla Inviato: lunedì 17 maggio 2010 12:01 A:

[asterisk-users] SIP Dialplan Failover Solution

Hello list, I need a hand to find the best dialplan failover solution when using two SIP Trunks. My reasons to do failover are: a) one of the two providers could be unreachable b) both providers may be UP but one of them could return a SIP error

[asterisk-users] Strange initial RING

Hello list! I'm having a strange problem with the VoIP Gateway that I'm using to go on the PSTN: if the number on the other end is busy or unavailable I hear an initial RING, generated by Asterisk from what I'm seeing and after that the line goes down with busy signal: Here is

[asterisk-users] R: Strange initial RING

Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] Strange initial RING On Fri, 19 Mar 2010, Alexandru Oniciuc wrote: Hello list! I'm having a strange problem with the VoIP Gateway that I'm using to go on the PSTN: if the number on the other end is busy

[asterisk-users] R: Asterisk hangup all incoming calls after 10 seconds

NOTICE[14413] rtp.c: Unknown RTP codec 126 received from '192.168.20.113' Maybe the codec 126 is the problem? [core] show codecs [core] show translation Maybe you don't have the codec required by your provider. Regards, Alex -Messaggio originale- Da:

[asterisk-users] R: Can not enable sip debug because CLI flooded

Edit logger.conf and set the desired log level. To disable the messages below just remove the severity notice from console. console = notice,warning,error,debug Alex Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di jonas kellens

[asterisk-users] t38 ATA

Hello, I need a hand in choosing a small ATA, even with one FXS port, that should do only fax with T38. I've tried Grandstream (ht286 model) but the faxes go out without ECM, even if the Fax machine has ECM enabled. Is there anyone that can

[asterisk-users] R: t38 ATA

On 03/13/2010 12:00 AM, Alexandru Oniciuc wrote: Hello, I need a hand in choosing a small ATA, even with one FXS port, that should do only fax with T38. I’ve tried Grandstream (ht286 model) but the faxes go out without ECM, even if the Fax machine has ECM enabled. Is there anyone that can

[asterisk-users] CDR duration/billsec

Hello list, I'm having troubles implementing the ${CDR(duration)} ${CDR(billsec)} variables in this scenario: PEER CALLS OUT - CALL GOES TO PEER'S DEFAULT OUTGOING CONTEXT - THE CALL IS SENT TO A MACRO AND GOES IN HANGUP - THE CALL RETURNS TO EXTENSION h OF

[asterisk-users] Empty SIP Packet

Hello list, debugging SIP, I found many empty lines like: --- SIP read from UDP://XXX.XXX.XXX.XXX:5060 --- - The IP address above corresponds to one of my accounts, which is behind a firewall. Is that normal, maybe some firewall that

[asterisk-users] Strange Problem

Hello list! I've run into a strange problem today and I was hoping that someone here has seen this before and maybe can give me a hand: I'm using asterisk 1.6.0.22 in this config: (A)PATTON ISDN -(B) ASTERISK - (C)PATTON PRI - PSTN - (D)OTHER PBX Strange Problem: USER A calls makes a call to

[asterisk-users] R: rtp.c:883 ast_rtcp_read: RTCP Read too short

The ring isn't fake :] The Linksys GW isn't dissing, is just responding to an INVITE. The problem is that you have problem passing voice. In other words: check RTP ports settings on server client or the firewall rules. Alex Da: asterisk-users-boun...@lists.digium.com

[asterisk-users] R: rtp.c:883 ast_rtcp_read: RTCP Read too short

Hello Wassim, server side you can check the RTP ports configured in rtp.conf which you will find in /etc/asterisk/. If the file isn't there, here are the defaults: ;[general] ; ; RTP start and RTP end configure start and end addresses ; ; Defaults are rtpstart=5000 and

[asterisk-users] R: How to dial multiple extensions at once likeinaring group and put them in conference?

Even though certain things should be discussed in private and that certain things should require a second thinking before stating them, I don't think you should impose a limit. Are you a Digium guy? An asterisk developer? Who are you? Apart the informational value of these lists I should hope

[asterisk-users] R: R: R: CDR(billsec)

Thank you! My bad,the CDR function was working on 1.4, I can confirm that endbeforehexten=yes does the trick, I've just tried it :] WOW: On 29/10/09 5:56 AM, Alexandru Oniciuc wrote: 12h difference! -Messaggio originale- Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk

[asterisk-users] R: R: R: R: CDR(billsec)

: R: R: CDR(billsec) On 30/10/09 1:10 AM, Alexandru Oniciuc wrote: Thank you! My bad,the CDR function was working on 1.4, I can confirm that endbeforehexten=yes does the trick, I've just tried it :] WOW: On 29/10/09 5:56 AM, Alexandru Oniciuc wrote: 12h difference! :D Yeah based in New

[asterisk-users] R: R: R: R: R: CDR(billsec)

Ye, don't mind that one ... Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Alexandru Oniciuc Inviato: giovedì 29 ottobre 2009 15.15 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: [asterisk-users] R: R: R: R: CDR

[asterisk-users] R: CDR(billsec)

Hello Anahi, I've encountered issues with CDR function when I was using the 1.4 version and was trying to get ${CDR(duration)} in extension h. Passing to 1.6.X.X resolved it. I hope this helps. Alex From:

[asterisk-users] R: R: CDR(billsec)

? *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Alexandru Oniciuc *Sent:* Wednesday, October 28, 2009 8:52 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk

[asterisk-users] Strange IAX2 / Iaxmodem problem

Hello. I'm having a strange problem with the IAX2 channel and IAXmodem and I was hoping to get some light from someone in these lists. On my logs and on the console I'm getting a bunch of lines with: [Oct 23 14:26:18] NOTICE[4417] chan_iax2.c: Peer 'XXX' is now UNREACHABLE!