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I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:

...:4569 After 5s: May 21 17:24:41 NOTICE[1133742896]: chan_iax2.c:5035 iax2_poke_noanswer: Peer '2004' is now UNREACHABLE! Some times: May 21 17:54:36 NOTICE[1133742896]: chan_iax2.c:5056 iax2_poke_peer: Still have a callno... Even checked the IAX peers, and nope, wasn't registered. SAMPLANET1*CLI> iax2 show peers Name/Username Host Mask Port Status 2004/2004 192.168.199.69 (D) 255.255.255.255 4569 UNREACHABLE My iax.conf is: [general] bindaddr=0.0.0.0 disallow=all allow=ilbc allow=gsm allow=ulaw allow=alaw bandwidth=low tos=reli...

What's the importance of the impedance matching in a FXO interface ? Kind regards, Miguel

-----Mensagem original----- De: miguel@amplanet.com.br [mailto:miguel@amplanet.com.br] Enviada em: quarta-feira, 11 de agosto de 2004 16:31 Para: 'asterisk-users-admin@lists.digium.com' Assunto: Asterisk --> Mediatrix 1204 --> returned -1: Operation not permitted (tethereal) Capturing on eth1 2.738748 192.168.199.4 -> 192.16...

...me type 1024, while native formats is 16 (read/write = 4/1024) == Spawn extension (from-sip, 23, 1) exited non-zero on 'SIP/2007-41d0' > -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of miguel@amplanet.com.br > Sent: Friday, July 09, 2004 11:31 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] ATA 186, firmware SIP 3.1 and codec g.726 > > I have a ATA 186 with SIP firmware 3.1 when I changed the configurations > to > use the g.726 codec I received many erro...

I have a problem: When I'm in a call and a second call arrive (call waiting) I can't transfer the first call. If I press flash the line change to the second call, if I press flash again the line change to the first call. How I can transfer a call in this kind of situation ? Kind regards, Miguel

>Message: 3 >Date: Thu, 26 May 2005 23:01:28 -0300 >From: "miguel" <miguel@amplanet.com.br> >Subject: [Asterisk-Users] AreskiCC >To: <asterisk-users@lists.digium.com> >Message-ID: <20050527020659.6171A2C001F@postfix.local> >Content-Type: text/plain; charset="us-ascii" >I'm tring to dowload the AreskiCC but the >www.areski.net is out...

As Promised, I've released a new version of Firefly (ver 1.8) with IAX & SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user -> software -> firefly), then delete tree from your registry. If that fixes it, send

I have two contexts and there I have some sip clients and some iax clients, in the sip clients a have extentions like, 20, 21, 22, 23, etc; in the iax clients I have some extentions like 2000, 2001, 2002, 2003, etc. My extention is 2003 when I make a call the manager program show me that the extention 20 was originating the call (in red, and show my name/extention)! When I make a call to

We are writing a program using the manager for * for our receptionist to use once the system go live. If anyone is interested in helping us with testing please let me know. We are designing it for a touch screen monitor for her to do transfers, see whose on the phone and a few other features. Its in the development stage and has bugs. but I think its gonna be really good. If your interested

I have a * server that receive calls from the internet interface (eth0, external) and from the local interface (eth1, internal), when the calls are between sip phones from the local interface (internal) all is ok, no echo, but when the call is between a external sip phone and a internal sip phone there are echo, a delay of 500ms, only on the internal side, the external side don't hear the

I have the * server in a machine with two interfaces one internal (eth1) and another external (eth0), I'm in the internal side and when I put a call to another internal sip phone all is ok, but when I put a call to a external sip phone I hear the echo of my voice, a delay of 500ms. Is here anyone that knows why ? Thank's a lot. Miguel

I have the user manual, I'll send it to your email tonight when I'll be in my home. I have an APA III-4FXO too, until today I can't put it to work with asterisk. Kind regards, Miguel Date: Fri, 03 Sep 2004 16:07:59 +1000 From: Jamie Carl <geek@j-code.net> Subject: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual? To:

Miguel, How many simultaneous calls (incoming and outgoing) you did using this implementation ? Kind regards, Miguel Date: Sun, 3 Oct 2004 11:21:04 -0500 From: Miguel Cavazos <miguel@cavazos.com.mx> Subject: [Asterisk-Users] Working E1 MFC/R2 M?xico !!! To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID:

On Ganeral --> Language correct from "portugese" to "portuguese". Kind regards, Miguel Date: Tue, 5 Oct 2004 09:47:08 +0100 (BST) From: gARetH baBB <hick.asterisk@gink.org> Subject: Re: [Asterisk-Users] Firefly 1.9.5 released To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID:

Where I can find the firmware for the Wellgate 3804 ? The files are: - 2m4sipfxo.103 - 4fxosip.103 I don't have a password to pick up it at the welltech site. Kind regards, Miguel

I have a channel bank (TA750) and a PRI with 30 channels connected to a TE405P, in the channel bank I have a extension to a fax machine, but it doesn't work to send or receive fax. There are any advice ? Kind regards, Miguel

When I try to make a call using the Mediatrix 1204 is showed on the CLI: -- Executing SetCIDNum("SIP/2009-4df1", "1111") in new stack -- Executing Dial("SIP/2009-4df1", "SIP/2217008@192.168.199.5") in new stack Aug 11 15:14:10 WARNING[1211108144]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x81 40c5c (len 794) to 192.168.199.5 returned -1: Operation not

Miguel, How many simultaneous calls (incoming and outgoing) you did ? Kind regards, Miguel Date: Sun, 3 Oct 2004 11:21:04 -0500 From: Miguel Cavazos <miguel@cavazos.com.mx> Subject: [Asterisk-Users] Working E1 MFC/R2 M?xico !!! To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID:

Gonzalo, I have an APA III-4FXO and I tried using your configurations, I received the message below: -- Executing Dial("SIP/2010-edfc", "SIP/2217008@Mediatrix") in new stack Sep 6 16:54:51 WARNING[1192491824]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x814bf0c (len 774) to 192.168.199.5 returned -1: Operation not permitted -- Called 2217008@Mediatrix Sep 6 16:54:54