search for: cerrajetto

...stem does not have the problem you describe. I can call from a SIP softphone, through asterisk , through the cisco and out to our meridian system or the PSTN. In fact, it works very well. Are you sure that you have the dial-peers on the router configured correctly? -----Original Message----- From: Cerrajetto [mailto:cerrajetto@pyme.net] Sent: 31 July 2003 09:09 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RTP codec 13 received - Cisco incompatibility? Hello, In our SIP network, Asterisk is the central PBX, and it routes calls to the PSTN thru a Cisco Router - IOS 12.2(11)T9. If a...

...e, sip:1234@myserver.com;extra_header=Uui?Uui=Hello Does Asterisk support this format? Is there a way to retrieve the value of these additional headers, and then decide the action (Dial, Playback, etc.) as desired? (I'm thinking in an "if ... then ... elsif ... else ..."). Thanks, Cerrajetto

...nd to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls of not registered phones ;Voz [541160] type=friend secret= insecure=no host=192.168.200.160 dtmfmode=inband ?Is there a way to indicate to DO NOT USE any password in the registration process? Thank you, Cerrajetto

...-- Attempting native bridge of SIP/pingtel01-af0d and SIP/192.168.200.200- a3d2 -- Got SIP response 400 "Bad Request - 'Invalid IP Address'" back from 192.168.200.99 == Spawn extension (default, 003, 1) exited non-zero on 'SIP/peter-af0d' Thank you very much, Mark Cerrajetto.

...rameters in URIs: exten => 1,1,Setvar,VXML_URL=var1=value1 exten => 1,2,Dial,sip/192.168.0.1 But, can I insert directly custom headers? Maybe I'm confused and Asterisk cannot do what I need ... Then, I would appreciate what could be the good direction. Thank you very much, Mark Cerrajetto.

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client

Hello! Is there a way to communicate from Asterisk to a TTS server? I've seen festival.conf, but it seems that it works only with Festival server. Thank you.

...: Sat, 13 Sep 2003 16:32:32 +0300 > From: Michael Manousos <manousos@inaccessnetworks.com> > Organization: inAccess Networks > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Asterisk using a h323 gateway > Reply-To: asterisk-users@lists.digium.com > > Cerrajetto wrote: > > Hello: > > > > I am testing Asterisk with oh323. > > > > My question is: can Asterisk route some calls thru a second h323 gateway (a > > h323 <-> PSTN gw)? > > > > - Asterisk ip: 192.168.1.10 > > - h323<->PSTN gw: 192....

Hello, Do you know a **real** sip fax server to _send_ faxes with Asterisk?. By example: an employee sends a TIFF facsimile by email to the SIP fax server; SIP fax server uses SIP to communicate with Asterisk; Asterisk communicates with a Cisco SIP2PSTN Gateway via SIP; Cisco sends the fax to the PSTN. I've read some posts than mention Hylafax, but it needs for real modems/analogic lines: