search for: pstn

I am new to Asterisk and the protocol/language complex world of VoIp and PBX. But I have a dedicated machine running A@H 2.8, a single TDM400P with one FXS module card connected to a standard analog phone. The second card is an X100P connected to my analog PSTN phone line. I also have Grandsteam IP phone plugged into the network and a couple of x-lite SIP softphones. I can make outgoing calls on the Grandstream or any registered SIP sofware phone from any computer. I can also get a dial tone from the analog phone connected to the ZAP X100P port. But when...

On 02/15/2018 03:44 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 6:43 PM, thelma at sys-concept.com wrote: >> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports >> >> IN audocodes setting I have: >> "EndPoint Phone Number" >> >> Channel: 3 phone number: pstn-4444 >> Channel: 4 phone number: pstn-9998 >> >> When I am calling " pstn-4444" the port number "Channel:3" lig...

On 02/15/2018 04:49 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 7:46 PM, thelma at sys-concept.com wrote: > > <snip> > >> >> Thanks again for the hint. >> Here is the output from asterisk. >> >> The call is coming on Audocodes gateway from: pstn-4444 >> >> But asterisk display: >> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060 >> >> Why not loolking up "pstn-4444" in sip.conf? > > It found pstn-4444 using 10.10.0.8:5060 - if the request always comes from the same I...

I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports IN audocodes setting I have: "EndPoint Phone Number" Channel: 3 phone number: pstn-4444 Channel: 4 phone number: pstn-9998 When I am calling " pstn-4444" the port number "Channel:3" lights up but asterisk is showing that the call is coming on "pstn-...

hi, i have a pronlem connecting call from pstn with the following confuguration, please advice extensions.conf [from-trunk] include => from-pstn [from-pstn] include => from-pstn-custom include => ext-did include => from-pstn-timecheck exten => fax,1,Goto(ext-fax,in_fax,1) extensions_custom.conf [from-pstn-custom] exten =&g...

...Colp wrote: > On Thu, Feb 15, 2018, at 7:03 PM, thelma at sys-concept.com wrote: >> On 02/15/2018 03:44 PM, Joshua Colp wrote: >>> On Thu, Feb 15, 2018, at 6:43 PM, thelma at sys-concept.com wrote: >>>> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports >>>> >>>> IN audocodes setting I have: >>>> "EndPoint Phone Number" >>>> >>>> Channel: 3 phone number: pstn-4444 >>>> Channel: 4 phone number: pstn-9998 >>>> >>>> When I am calling...

Hello list, this is my first post on this list. I have a server with Asterisk and a Linksys SPA3102 with 3 SIP phones. I have configured the SPA PSTN line as trunk to receive and send calls. I can call outside from SIP phone throw the PSTN line and all is OK, the problem is when I receive a call from the PSTN, on the out caller phone there is a demo playback. I want to redirect the call to a internal SIP phone. This is the extensions.conf: [s...

Hi, all I am trying to figure out how to forward incoming PSTN call on SPA3000 to VoIP extension(s). Basically, I have converted my home to VoIP. I have normal phone (connected to SPA3000) and couple of IP phones. All call coming from VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I need to do same thing for incoming PSTN calls. I have en...

Hi, I'm trying to transfer an incoming call from the PSTN to another PSTN number through a SER - Asterisk system. SER doing only routing.. pstn call-> SER -> asterisk (call forward) -> SER -> pstn Logic for SER: If something comes from the pstn, send it to asterisk. If something comes from asterisk, send it to the pstn. Every time I am g...

This post is simply documenting a spouse-friendly way of using the spa-3000 as both a fxs and fxo port for basic soho environments in the US, allowing asterisk to participate as needed/wanted. All home phones are connected _only_ to the spa-3000 fxs port. The incoming home pstn line is connected _only_ to the spa-3000 fxo port. Defined Line 1 (fxs) to register with asterisk via sip (extn 1111), with silence suppression disabled. Defined the PSTN Line (fxo) to register with asterisk via sip using a second sip.conf entry (extn 2222). PSTN User, defined PSTN Ring Thru...

I think I have it configured properly. ztcfg -vv shows it as channel 1 and zttool shows it as OK. But I can't dial out. When I try, it shows it arrive in teh right stack, but then issues the following errors: channel.c:1676 ast_request: No channel type registered for '{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}' = = Everyone is busy at this time My config files are below: sip.conf [general] port = 5060 bindaddr = 0.0.0.0 disallow=all allow=gsm allow=ulaw allow=alaw allow=G723.1 context=from-sip [2000] type=friend use...

...type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) dahdi seems up. I restarted. Rebooted. Now I've reverted to 1.4. CLI> dahdi show channels Chan Extension Context Language MOH Interpret Blocked State pseudo from-pstn en default In Service 1 from-pstn en default In Service 2 from-pstn en default In Service 3 from-pstn en default In Service 4 from-pstn...

We manage our own VOIP solution using Asterisk. Has anyone had success with an IP-PSTN provider? I'm looking for someone to terminate SIP calls to the PSTN in the Seattle, Washington area. (vice-versa as well if possible) Yes, I could do it myself via asterisk and digium cards but I would like to consider other options. Any opinions? Thanks, Chad -------------- next part...

Hi all, I have my asterisk setup as following: IP 2 x E1 x-lite <-------> Asterisk -------> PSTN When I place a call from x-lite to PSTN, the quality of the sound in the direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user, heard by the PSTN user is choppy and makes communication not very pleasant. The sound is choppy as if bits of data were lost. The strange thing...

...pent two days trying to solve this issue but to no prevail and I'm hoping to get some help. I've configured Asterisk as a SIP client, running on OpenWRT on an embedded device with onboard FXS and ATA. Asterisk is connecting to an external SIP provider on the Internet who in turn provides a PSTN gateway. I'm able to make calls to other SIP accounts registered on the same server who are outside my LAN. However, I can not make calls to any PSTN numbers. When trying to make PSTN calls it sounds like the person at the other end is immediately rejecting the call although I know this is not...

* --> SIP --> CISCO --> PRI --> PSTN The PSTN sees no callerid. *---> PRI[zaptel]--> PSTN Callerid is there... which makes me think it's the cisco, not the PRI/PSTN/telco. CISCO PRI--> * PRI [zaptel] Callerid IS there... which makes me shake my head in disbelief, because * can see clid from the cisco pri, but pstn do...

...VERBOSE[4526] logger.c: -- Accepting overlap call from '0339' to '<unspecified>' on channel 0/31, span 1 [Feb 11 17:45:25] VERBOSE[5764] logger.c: -- Starting simple switch on 'Zap/31-1' [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [91905 at from-pstn:1] Set("Zap/31-1", "__FROM_DID=91905") in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [91905 at from-pstn:2] NoOp("Zap/31-1", "Received an unknown call with DID set to 91905") in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c:...

Hi This is the output from show dialplan dial-sipmnf-sippt-pstn [ Context 'dial-sipmnf-sippt-pstn' created by 'pbx_config' ] 's' => 1. Verbose(1,Dialing ${ARG1} on mnf pt pstn) [pbx_config] 2. Dial(SIP/${ARG1}@${SIPMNF},${ARG2},${OUTBDIAL}) [pbx_config] 3. Set(GLOBAL(FOUNDME)=${DI...

Hi All After lots of try I was successfull in connecting to PSTN to make and recevice calls , I used AMP for this purpose , now I wanted to try out this Asterisk server answers the call , ask for the extensions and then after the extension entered the call is forwarded /transfered to the extension no , I use Asterisk 1.2.4, configured using AMP , on RHEL3...

Hello, I'm facing problem where as whenever there are incoming call from pstn, there will be no audio coming in. User at the other end also could not hear my voice. This happens few days back. Im using asterisk 1.6.1.2 with dahdi tool 2.2.0. I thought it was time to upgrade, so upgraded to dahdi 2.2.1 and asterisk 1.6.2.5. However, it does not help at all. My current confi...