Re: [asterisk-users] 10.2.0-rc2: permitted contact can't register.

You want to allow single IP or whole subnet ? Regards, Zohair Raza On Wed, Feb 29, 2012 at 4:44 AM, sean darcy wrote: > An outside device can't register: > > WARNING: getnameinfo(): ai_family not supported > WARNING: chan_sip.c:14456 parse_register_contact: Domain > '69.xxx.yyy.zzz:5060' di

Re: [asterisk-users] outbound fax over t38 gateway can't pass

btw, played with res_fax.conf if I set maxrate=7200 fax machines try (and fail) 9600 anyway. Why? If limited ti 7200? looks like bug... So I set maxrate=4800 and modems=v27. Faxes pass Looks like problems with V29... 29.02.2012 07:56, Dmitry Melekhov пишет: Hello! I have problems with o

Re: [asterisk-users] Alphanumeric DTMF !?

Eric thats really a nice idea to communicate between two or more of "our" servers. Make the call to the remote system and send the digits in the "encoded > string", you will need something on the other end to decode the text. But the other end is not our's but could be any solution which require

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Imass Sent: Tuesday, February 28, 2012 10:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Same provider

[asterisk-users] outbound fax over t38 gateway can't pass

Hello! I have problems with outbound faxes with asterisk 10.2 t38 gateway. There is asterisk box, connected to panasonic kx-td500 over PRI link with TE122. If we try to send fax with following path: panasonic 500 extension fax machine panasonic500-> asterisk-> ooh323-> cisco 3845-> fax mach

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

On Tue, Feb 28, 2012 at 8:28 PM, Carlos Alvarez wrote: > On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass wrote: >> Please expand as to how you set-up a SIP ATA behind a common home >> router set-up, without port redirection and/or use of a SIP proxy >> and/or STUN server? Unless the ATA has some

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

Just to stir the pot a bit, I am a member of a worldwide private network of Asterisk and AstLinux users. the network uses IAX exclusively, and we have no issues relating to audio quality with a large variety of providers, routers, host machines, and expertise in configuration of the specific no

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

Roger That! On Tue, Feb 28, 2012 at 8:28 PM, Carlos Alvarez wrote: > On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass wrote: > > Please expand as to how you set-up a SIP ATA behind a common home > > router set-up, without port redirection and/or use of a SIP proxy > > and/or STUN server? Unless

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

On Tue, Feb 28, 2012 at 7:41 PM, Carlos Alvarez wrote: > On Tue, Feb 28, 2012 at 5:35 PM, Alejandro Imass wrote: > > works. This cannot be done with SIP and off the shelf cheap ATAs, > > period. > > We do it, so "cannot" seems to be a strong word. It's not perfect, > but our IAX problems outnumb

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

On Tue, Feb 28, 2012 at 7:07 PM, Alejandro Imass wrote: > On Tue, Feb 28, 2012 at 6:36 PM, Steve Totaro > wrote: > > [...] > > > Without trunking, you only have the single port thing. It is quite easy > to > > Nope. The main reason _we_ use IAX is because it's easier for NAT > > And it is easie

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imass wrote: > Please expand as to how you set-up a SIP ATA behind a common home > router set-up, without port redirection and/or use of a SIP proxy > and/or STUN server? Unless the ATA has some sort of magic (e.g. VPN > support) it _cannot_ be done. Go

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

On Tue, Feb 28, 2012 at 7:41 PM, Carlos Alvarez wrote: > On Tue, Feb 28, 2012 at 5:35 PM, Alejandro Imass wrote: >> works. This cannot be done with SIP and off the shelf cheap ATAs, >> period. > > We do it, so "cannot" seems to be a strong word.  It's not perfect, Please expand as to how you set

[asterisk-users] 10.2.0-rc2: permitted contact can't register.

An outside device can't register: WARNING: getnameinfo(): ai_family not supported WARNING: chan_sip.c:14456 parse_register_contact: Domain '69.xxx.yyy.zzz:5060' disallowed by contact ACL (violating IP ) sip.conf: [general] ... alwaysreject=yes dynamic_exclude_static = yes allowguest=no con

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

On Tue, Feb 28, 2012 at 5:35 PM, Alejandro Imass wrote: > works. This cannot be done with SIP and off the shelf cheap ATAs, > period. We do it, so "cannot" seems to be a strong word. It's not perfect, but our IAX problems outnumbered the SIP problems by at least double. Your mileage clearly vari

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

On Tue, Feb 28, 2012 at 7:19 PM, Carlos Alvarez wrote: > Perhaps your users live in an internet ghetto where the routers are > similar to Yugos with spinners.  We haven't run into any routers that > don't do NAT properly in a very very long time. > Perhaps you should read again and point out wher

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

Perhaps your users live in an internet ghetto where the routers are similar to Yugos with spinners. We haven't run into any routers that don't do NAT properly in a very very long time. On Tue, Feb 28, 2012 at 5:07 PM, Alejandro Imass wrote: > On Tue, Feb 28, 2012 at 6:36 PM, Steve Totaro > wro

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

On 2012-02-28 23:29:53 +, Steve Totaro said: They said the same thing in 2005, 2008, now Every release. You never answered the question as to why you don't want to use SIP. Is there a reason, or do you just want to torture yourself? Probably self-torture, yes. I want to at least try

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

Oops, I meant da Asterisk 'hood. Thanks for the protip. On 02/28/2012 06:55 PM, Steve Totaro wrote: Hey Alex, Hope you are well. Just a piece of advice. Many or most people do not know the real definition of ghetto and take it as a negative, poor, racial, black, connotation. Your vocabular

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

On Tue, Feb 28, 2012 at 6:36 PM, Steve Totaro wrote: [...] > Without trunking, you only have the single port thing.  It is quite easy to Nope. The main reason _we_ use IAX is because it's easier for NAT > open the correct ports for SIP, some just have GUI with a SIP checkbox, It may be true f

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

Wow Wikipedia was the only place that had the original meaning and not the slur or slang meaning. " A *ghetto* is a section of a city predominantly occupied by a group who live there, especially because of social, economic, or legal issues. The term was originally used in Venice

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

Hey Alex, Hope you are well. Just a piece of advice. Many or most people do not know the real definition of ghetto and take it as a negative, poor, racial, black, connotation. Your vocabulary and and ability to articulate correctly can get you in trouble sometimes. Anyone that thinks that the

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

IAX is not supported or taken seriously outside the Asterisk ghetto, and that's good enough reason not to use it, IMHO. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

BTW, Trunking was the other selling point of IAX2 besides using 1 port which is easily a DDOS target and also probably still an implantation problem of using one thread and one proc for all calls. Trunking allowed for less overhead then SIP since all the overhead for the concurrent calls were comb

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

They said the same thing in 2005, 2008, now Every release. You never answered the question as to why you don't want to use SIP. Is there a reason, or do you just want to torture yourself? Thanks, Steve T On Tue, Feb 28, 2012 at 6:23 PM, Troy Telford wrote: > On 2012-02-28 21:22:44 +,

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

On 2012-02-28 21:22:44 +, Kevin P. Fleming said: A serious bug with IAX2 trunking in recent versions of Asterisk (you did not mention what version you are using) was just resolved last week. You should test with 'trunk=no' to see if that is the cause of your problem; it seems very likely.

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

And the dude arrives talking about penis.. On Tue, Feb 28, 2012 at 6:07 PM, Carlos Alvarez wrote: > I have no interest in the penis-measurement competition firing up > here, but I'll say that we have 100% abandoned IAX from all of our > systems due to a myriad of issues. These days it offers

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

Roger That, I am an IC. I contract with the Government to little ten phone shops. From VA/MD/DC area, I have been contracted and flown in to many large call center locations that were CONUS and OCONUS. My facebook is Steve Totaro in Reston VA. LinkedIN is more accurate, but my resume speaks the

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

I have no interest in the penis-measurement competition firing up here, but I'll say that we have 100% abandoned IAX from all of our systems due to a myriad of issues. These days it offers no real advantages in our opinion. On Tue, Feb 28, 2012 at 4:03 PM, Steve Totaro wrote: > People around he

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

People around here either hate me or love me. I post experience and am accused of bragging or whatever. As a reader, I want to know who is giving me advice and what it is based on. $40k/wk of long distance through VoicePulse. I have the invoices, that is high usage, others attack me for posting

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

On 2012-02-28 22:17:37 +, Danny Nicholas said: Ok Steve, obviously you've outsmarted at least this poster.  On the one hand, IAX2 has purchased things for you (won't go as far as saying it bought your Mercedes), but on the other hand it is being dropped by providers as we speak. So are you

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

encryption=yes is meaningless if the provider doesn't support it (mine doesn't). I put it there in the wild hope they eventually will - and no config change will be needed on my part. Still, when I changed it to encryption=no, and tested there wasn't any difference. So I've tried disabling t

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

I've tried turning jitterbuffer off - doesn't make a difference. (And why should it? The Jitterbuffer only applies to incoming calls, doesn't it?) On 2012-02-28 21:12:48 +, Noah Engelberth said: I'd try turning off the jitterbuffer and see if that makes things better. I just traced a sim

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

On 02/28/2012 03:08 PM, Troy Telford wrote: [myprovider] type=friend username= secret= context=somecontext host=provider_server qualify=1000 disallow=all allow=g729 allow=ulaw auth=md5,rsa requirecalltoken=yes trunk=yes A serious bug with IAX2 trunking in recent versions of Asterisk (you did

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

My first two guesses are that encryption is hosing you or that the "single-channel" nature of IAX2 may have something to do with it. IAX2 "talks" on 1 channel, SIP uses "twisted pair" connotation on two channels (as I understand it). -Original Message- From: asterisk-users-boun...@lists.d

Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

I'd try turning off the jitterbuffer and see if that makes things better. I just traced a similar call quality issue transferring calls incoming DAHDI on one * box to another * box, and turning off the jitterbuffer on the side that "couldn't hear" (in my case, the * box with the DAHDI lines, as

[asterisk-users] Same provider - IAX sounds bad, SIP sounds great

On my Asterisk system, I'm using a provider that provides both IAX2 and SIP connectivity. Personally, I'd prefer to use IAX2, and that's what my account is setup to use. However, I'm having a problem: With IAX2: - Incoming Voice from my Provider -> Asterisk = Sounds great - Outgoing Voice fro

Re: [asterisk-users] better timing source for an asterisk gateway

On 02/28/2012 09:06 AM, ml asterisk wrote: Hi, I have to make an asterisk gateway in front of several other asterisk. This gateway will essentialy be used for outbound call. This gateway will be connected to other asterisk by IAX trunk, outbound call will use SIP trunk (voip provider or patton i

Re: [asterisk-users] Alphanumeric DTMF !?

Just for fun I did something similar at one point. 0-9 A-D and * and # make a character set of 16 characters, perfect for encoding as hex. Take your string, get the ASCII value of each character, convert it to hex, and add it to the "encoded" string. Just before dialing, replace all "e" with #

Re: [asterisk-users] Asterisk auto-dial out a SIP .

On Tuesday 28 February 2012, upendra wrote: > hi, > > an anyone tell me how to do auto dial to a SIP in Asterisk using a script > .any example will be more helpful ... ! > > > Regards > Upendra You need to inject a callfile into /var/spool/asterisk/outgoing . The file should look something

[asterisk-users] better timing source for an asterisk gateway

Hi, I have to make an asterisk gateway in front of several other asterisk. This gateway will essentialy be used for outbound call. This gateway will be connected to other asterisk by IAX trunk, outbound call will use SIP trunk (voip provider or patton isdn). I have a TE220BF available than i ca

Re: [asterisk-users] Asterisk Version 1.8.9.2 Question About SIP/SRTP/TLS

On 28/02/2012 7:58 PM, DHAVAL INDRODIYA wrote: Hi All, I have one question that if my device is registered over TLS on asterisk . is it required that it can only use SRTP for making an outbound calls or incoming calls too. No. how we can disable srtp and only enable TLS. tlsenable=y

Re: [asterisk-users] TCE400P diagnostic messages

On 02/27/2012 06:00 PM, Chris Sohns wrote: Hello, I’m running Asterisk C.3.7.2 and DAHDI 2.4.1.2 No changes in hardware since last April and asterisk version since last summer. Since last Friday one of the call centers I support has been complaining of lagged audio, dropped calls, and “two cal

Re: [asterisk-users] Asterisk auto-dial out a SIP .

Depends on your coding preference. you need to put the files in your spool directory (IE /var/spool/asterisk/outgoing) Just search google for asterisk call files. should help you out ;) On 12-02-28 07:59 AM, upendra wrote: hi, an anyone tell me how to do auto dial to a SIP in Asterisk using

Re: [asterisk-users] asterisk MCID detection

On 28-02-12 13:52, Deka, Rajib IN MAA SL wrote: Hello List, Does asterisk 1.8 or 10 provides any events or Dialplan application to detect MCID (Malicious Caller Identification)in an incoming call? For those unfamiliar with MCID: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/4_0_1/c

[asterisk-users] Asterisk auto-dial out a SIP .

hi, an anyone tell me how to do auto dial to a SIP in Asterisk using a script .any example will be more helpful ... ! Regards Upendra -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk?

[asterisk-users] asterisk MCID detection

Hello List, Does asterisk 1.8 or 10 provides any events or Dialplan application to detect MCID (Malicious Caller Identification) in an incoming call? Please provide any sample Dialplan if possible. Thanks & Regards, Rajib -- _

[asterisk-users] Asterisk Version 1.8.9.2 Question About SIP/SRTP/TLS

Hi All, I have one question that if my device is registered over TLS on asterisk . is it required that it can only use SRTP for making an outbound calls or incoming calls too. how we can disable srtp and only enable TLS. is there any dial-plan functions that can help to disable/enable this SRTP