search for: ptsn

...show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help or any cluees? Verbosity was 5 and is now 7 -- Starting simple switch on 'Zap/1-1' == Unknown extension '8' in context 'from-ptsn' requested -- <Zap/1-1> Playing 'ss-noservice' (language 'en') -- Hungup 'Zap/1-1' ivr-server*CLI> ivr-server*CLI> dialplan show [ Context 'defaults' created by 'pbx_config' ] Include => 'from-ptsn'...

Hi all, I have setup a Dell PowerEdge 2550 with a Sangoma A200 card with 2xFSO and 1XFS modules. The PowerEdge specs are 1 x P3 1133MHz, 512MB RAM. Sangoma A200 has 3 analogue PSTN lines connected. This server is based in Office 1, with 5 users all with a Linksys SPA942 VoIP Handset. There is another Office (Office 2) connected to here using VPN. There are two users in Office 2 with the

I've been looking for a good iax or sip <==> ptsn provider. Someone with very low cost usa calling and can offer incoming ptsn connections in most markets. The only decent providers I could find were iconnecthere and nufone. Has anyone found someone that really stood out? Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Av...

...-admin@lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > > 1. RE: voicemail file access problems (Todd Lieberman) > 2. sip -> h323 -> ptsn (Brian West) > 3. RE: voicemail file access problems (Todd Lieberman) > 4. Re: voicemail file access problems (Tilghman Lesher) > 5. Re: sip -> h323 -> ptsn (Patrick) > 6. RE: voicemail file access problems (Patrick) > 7. Re: sip -> h323 -> ptsn (Brian West...

...messed me up. This is all running on Ubuntu Server 12.04. I have been googling/researching reading the book, etc. Everything I find is for SIP softphones etc. I just want to start by getting the asterisk machine to provide dialtone to the analog phone, and ring that phone when I call the PTSN line. I must be missing something in the basic dahdi and dialplan to simple get the analog phone to work. Can someone point me to a example of what I am trying to accomplish? Not wanting handholding but a push in the right direction. Thanks. -------------- next part -------------- An HTML att...

Good morning, I've been having some problems getting the SPA-3102 working properly in India. Specific problem is that calls from the Asterisk server out the FXS port is failing. When trying to make calls, I'm getting this message: [Jan 26 23:00:31] NOTICE[2136]: chan_sip.c:13774 handle_request_invite: Call from '' to extension '66200' rejected because extension not found.

...has 1 FXO port) It working ok , but the device SPA3102 do this : when a call is placed for outgoing in asterisk and send to SPA3102 , this device "answer and dial the number in the same time" , in my CLI I see the channel is open , but on the phone I hear the ringing from the provider PTSN , then the answer.... , So , in the end, asterisk don't know when the real answer was made on PTSN line. It like SPA3102 don't notify asterisk the ringing , then open the channel. There is a problem for signaling ? Sincerely , Alexandru Achim

Hi All, I Have a Digium Card TDM40P, the specification say: OEM TDM40B: TDM400P + 4 PORT FXS Bundle, my question is: Can I to install analog lines of PTSN?, other detail is: this card have 4 card green. I need to know what is the best card for the following scenario: I need a IVR for my comapny and a PBX, but i want that my extension not use FXS I want IP phone . Thanks ins advanced, Regards, ___________________________________________________...

I have a tdm 3xfxs and 1xfxo, aslo I have a setting with 1 snom 190 and 2 xten line. I can call from the snom to the ptsn line at the fxo port ok. I can call from the ptsn to the xten lite phone. I can call from the xten lite to snom but what I CAN`T do is; Call from xten to ptsn. When I dial from the xten, I can hear the dialed party, but he cannot hear me... Tips? Help? What I'm doing wrong? TIA

Hello Asterisk List, My company is running a bunch of Asterisk servers behind a Kamailio (openSER) SIP proxy gateway. Calls come in from our PTSN to VOIP service to Kamailio, which then randomly chooses an Asterisk server to handle the call. All Asterisk servers are 1.6.0.9, but the issue I'm about to describe exists in 1.6.1.5-rc1 as well. Ultimately what I want to do is cap each Asterisk server at a maximum number of simultan...

Hi all. I`ve installed a X100P on my box and is working well with incoming and outgoing calls as a trunk with one PTSN line. I want to connect the X100P to my Panasonic 616 EASA-PHONE as an internal extension to permit to users to make calls to SIP devices from analog phones, the problem is when I dial the ext number where the X100P is connected I get busy tone. What config I need to change to my asterisk files t...

...pdated conferencing on 73, with 0 conference users Nov 12 12:48:56 VERBOSE[32609] logger.c: -- Hungup 'Zap/73-1' Nov 12 12:49:03 DEBUG[32571] chan_zap.c: disabled echo cancellation on channel 73 extensions.conf follows: ; --- First all the incoming routes --- ; from outside T1 [from-ptsn] exten => s,1,Answer() include => intertel-ext exten => t,1,Playback(vm-goodbye) exten => t,2,Hangup() ; from intertel-axxess box [from-intertel] include => internal ; generic interal route [internal] exten => s,1,Answer() include => intertel-ext include => to-ptsn ; ---...

Alright I've been having an issue when trying to dial out locally when coming from SIP. This used to work no problem, now it doesn't. Now the local PRI to Bell Is working fine I have calls coming in and out of it constantly right now. BUT if I try and make a local call from SIP (from X-Lite or one of our Linksys SPA2102s) It fails every time with errors like these == Using SIP RTP

...d pick up a SIP device to talk to Asterisk, and configure the PBX to use packet8 for all US calling. Being new to this, just want to know if this is a reasonable approach or totally newbian. Also, I saw nufone was very Asterisk friendly, do they support inbound calls, meaning can they assign me a PTSN number, or is their service for outbound calling only? Thanks

I'm confused a bit about the following and was hoping to get some answers on this group - What is exactly implied when we say asterisk can connect to a PSTN. Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume asterisk does not need to do any SS7 signaling and all it does (playing the role of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct statement?

...Currently I'm playing with DISA, but I'm worried this will happen when I get to implementing AAs etc. I have a free SIP trunk from IPKall that I'm trying to make work. I'm able to receive calls, and I've now setup and extension with DISA and a password. I connect ok from the PTSN, get the dialtone, and enter the password. In the CLI I'm getting duplicate/extra/incorrect digits. I've tried dtmfmode=auto, dtmfmode=inband, and dtmfmode=rfc2833 all with similar results. For testing I set the password to 67891 here's what I'm getting (small sample size, but it...

...local analog cordless phone, a local PSTN line and the setup to link to an asterisk server located at a remote static ip address. I can dial the cordless phone from other extensions located at the asterisk server; I can dial out from the cordless phone trough the Sipura - Asterisk link, using the PTSN line on the other port of the Sipura. So far, so good. BUT: While I can receive a phone call arriving on the PSTN port, it is correctly routed to the cordless phone on the other spa port with the faked callerid trick found in the wiki, the spa does not seems to detect the end of the call. So aft...

This indicate that the connection with the local provider PTSN it is ok? : -- Attempting call on Zap/10 for s@inbound:1 (Retry 2) -- Making new call for cr 32781 > Protocol Discriminator: Q.931 (8) len=28 > Call Ref: len= 2 (reference 13/0xD) (Originator) > Message type: SETUP (5) > Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info t...

Hi, I am new in asterisk. Is it possible to use it with common dialup modem to connect ptsn to the server? Thanks Regards, Soragan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040123/8e77b5cf/attachment.htm

Hi, When someone calls into our * system over a PTSN line, we answer with a recorded prompt. (Thank you for calling, etc..) The first second of this prompt ALWAYS skips. After that, everything sounds great and works perfectly. There is nothing wrong with the prompt. Does anyobdy have any clues?? Thanks, -N