search for: telx

I am trying to send calls from (telx-NY17S) to (telx-nyc) via an IAX2 channel. However the call is being rejected on the (telx-nyc) server. See error below copied from telx-nyc CLI> Apr 22 13:56:57 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected connect attempt from 192.168.0.251 I have icluded the following con...

To All I am using a SNOM 190 w/Asterisk server. Here is my sip.conf [7501] type=friend context=external username=7501 callerid="Telx 7501" <7501> mailbox=7501@telx.com host=dynamic dtmfmode=rfc2833 My question is this. With above settings in my sip.conf specially "dtmfmode=rfc2833" What should my "DTMF Payload Type:" be set to on my SNOM 190 phone. Currently it is set to 101. Should it be s...

Good day all How do i set a connection between 2 asterisk servers via PRI In Bri I would set one to NT and TE How shoud the zapata.conf and zaptel.conf look And how should the cable be? All I got on the web was to set one to "pri_net"...this cant be all? And the cable > pin1 <--> pin4> pin2 <--> pin5> pin3 <--> pin6> pin4 <--> pin1> pin5 <-->

...nnected it required me to enter a 4 digit PIN. Now here is the problem whenever I enter a pin it is received twice. For example if the PIN is 1234 they receive it as 12341234. Any ideas what could be wrong? BTW we are using SNOM 190 ip phones (sip) Regards, Michael DiMartino Director of MIS The telx Group, Inc. 17 State St, 33rd Floor New York, NY 10004 T: 212.480.3300 X2022 C: 646.207.6603 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050114/6a85da1a/attachment.htm

...to provide remote SIP phones (home offices) securely. If the solution is to put up another Asterisk server with a public IP address I am opposed to that. I am looking for the a secure reliable solution to set up remote SIP phones. Thanks in Advance. Regards, Michael DiMartino Director of MIS The telx Group, Inc. 17 State St, 33rd Floor New York, NY 10004 T: 212.480.3300 X2022 C: 646.207.6603

Hello - sorry for my bad english. I will make a list of all sound files on asterisk and i'll record then on professional studio. the french prompts from sineapps sounds bad... sorry for her... tell me if their is many peoples want it ! thank's. en francais: dites moi si ca vaut le coup que j'investissexr dans l'enregistrement des messages en francais. La voix sera la "voix

I am looking for a good Asterisk GUI to manage my server. Any Suggestions? Regards, Michael DiMartino Director of MIS The telx Group, Inc. 17 State St, 33rd Floor New York, NY 10004 T: 212.480.3300 X2022 C: 646.207.6603 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041124/96bf8fea/attachment.htm

To All; My current issues is a 5 second delay for call that is being transferred from the Norstar units to the Asterisk servers VIA a PRI. Is their anything that can be done to speed up the transfer on the Norstar. Below is my current phone config. < Norstar1 >----PRI----< Asterisk-1 >----IP-WAN----< Asterisk-2 >---PRI---< Norstar2> The Norstars are MICS 0x32 4.1

...he official launch of a registry that allows service providers routing calls over the Internet to avoid paying local phone companies access charges. The VPF ENUM Registry allows carriers to map telephone numbers to IP addresses for such things as SIP phones and e-mail servers, Stealth announced at telx's Customer Business Exchange meeting in New York City. A phone call routed to a number listed in the registry would be terminated over the Internet, rather than over the traditional phone network controlled by the regional Bells and other local exchange carriers. The registry "has the pot...

Hi, Could somebody help me this SIP trasport? I'm receiving SIP "invite" with CLI of calling party from the SIP gateway, aster that my IVR has to answer the call. sip.conf: ========= [general] port = 5060 bindaddr = 0.0.0.0 context = incomingsip videosupport=yes ; Turn on support for SIP video disallow=all ; Disallow all codecs allow=g729

If you ring into * and leave voicemail It does not reset the line Any ideas would be appreciated Regards Mick

Hi, I haven't seen any mention of this on the list. I'm curious if anyone has tried it and can share some opinions on it? http://www.thirdlane.com/screenshots.htm http://www.thirdlane.com/opensource.htm#manager Defaults Manager - initial PBX configuration Device Manager - management of devices (phones) Mailbox Manager - configuration of user mailboxes Extensions Manager - dialplan

Hi, I wonder if you guys can recomend a good SIP phone. A phone thats works great with * has a lot of features, and is cheap. Actually all kind pf VoIP hardware is of interesst. Is there a really good site for VoIP harware ? /Mike

to start with i am new to asterisks and i am also a telcom idiot. with that said i have one vonage line i would like to hook up in my soon to be built Asterisk ippbx server. Now with the one Vonage (with call waiting) line can i receive more one call using an auto attendant route the call the approiate extention? thanks mike

I have the dtmfmode in sip.conf set to use rfc 2833 however, when my users have to enter pin numbers to join let say someone's conference bridge the pin is received twice. Any ideas on how to solve this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050120/56ff46fa/attachment.htm

What is the unsubscribed address? Thanks Michael -----Original Message----- From: Steve Underwood [mailto:steveu@coppice.org] Sent: Friday, February 18, 2005 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Re: quadbri and spandsp You need to use the caller parameter. Something like: Channel:Zap/G1/XXXXXXXX Application:txfax

I agree with that fact the same questions get posted, but that problem is compounded by the fact the archives are not really searchable. If the were as lease some users would search. The archives need to be fully indexed. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of steve szmidt Sent: Monday, June 27, 2005