similar to: Asterisk / 183 message

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?

Pardon my ignorance, but I was hoping someone could clear up something for me. - For a few POTS lines, digium has a single port card for that, or a T1 card to a channel bank. - For 10 or more lines, digium has a T1 or E1 card for that too based on PRI channels - For 100's to 1000's of lines, I suspect a soft-switch is in order??? A traditional phone company will sell: - POTS lines for

Hello! I migrated asterisk 11 to 13 as user of FreePBX 12.0.76.2. As customer of German Telekom, I have three numbers and therefore three trunks - each number is bound to one trunk. After migration, some callers complained about missing ringback tone: they didn't hear any ring tone and where surprised that they suddenly got me anyway :-). The connection afterwards was as expected. Deeper

Hi, I use in Production : Asterisk 1.2.9.1 We Use Asterisk as a SIP Transit Server to record centrally all the calls. The call flow would be: incoming calls : PSTN -> GW -SIP-> Asterisk(Record) -SIP-> Softswitch -> IP Phone outgoing calls : IP Phone -> Softswitch -SIP-> Asterisk(Record) -SIP-> GW -> PSTN Dial plan in Asterisk is quite simple: [record] exten =>

Here is a question that needs a few opinions... Recently we installed a couple of FXS gateways into a site with a SIP Proxy/Softswitch other than Asterisk. One of the things that the users on this site need to do is receive calls on single line phones on the FXS gateways and then hookflash and transfer them to other SIP users. We found that the FXS units, true to their nature as VoIP gateways,

Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system & restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository:

Got at suggestion from CarrierAccess to use the Adit 600 as an VoIP router using MGCP IP protocol, instead of controlling it through an E1. Have anyone tried this configuration? How does MGCP works? I've tried to search for it on Google, but I only find the protocol specification for it. Is Asterisk fully capable of this? I can't find any documentatin covering the use of MGCP in Asterisk.

Hello, We have an asterisk 1.2.13 box that use a Digium TE205P T1 PRI connect to the telco, users mainly use snom 320/300 SIP phones. When dialing to an external phone number with custom ringback music, users reported that they could not hear the music but can only hear the standard ring tone generated by the system. Is there any kind of settings need to allow the ringback music pass to the

Hi, I am using asterisk 1.8.22 and have a problem when calling in parallel several SIP endpoints and I am not sure how to resolve it. In this case Asterisk will not bridge any audio to the caller before the 200 OK. Which means any progress announcements, including remotely generated ringback, are not passed back to the caller. This behavior is completely correct, because there is no way to know

Has anyone had any success using the Adit 600 with the CMG card talking MGCP to asterisk? I want to have a central asterisk server with 10 Adit 600's at various locations providing 24 FXS ports.... Thanks, Isaac

Hi, Has anyone using chan_vpb noticed that only one splash of ringback is provided to the PSTN? I have tried several different permutations in extensions.conf and interworking to mgcp sip and iax. I am using the Voicetronix supplied chan_vpb and asterisk from the 1.0 cvs source tree. thanks darren -------------- next part -------------- An HTML attachment was scrubbed... URL:

I don't expect my SIP provider to provide useful "Remote-Party-ID" information. Therefore, I am using "CONNECTEDLINE(num)=xxx" AND "CONNECTEDLINE(name)=yyy" to populate remote party information from a local database. I am also using the "I" (upper case "i") option for Dial. Generally I like to see to see the remote party name appear on the

Hi, I picked up a router with 8 voice ports that supports MGCP, but it has several options that I am not familiar with or do not seem apparent in the mgcp.conf. Enter the default IP address for the Notified Entity: [0.0.0.0] Enter the listening port of the Notified Entity: [2427] Enter the IP address for MGCP signalling (Data): [192.168.0.210] Enter the local port for MGCP signaling (Data):

Dear all I have confusion how to asterisk genrate tone and what ringing code use default 180 or 183 i have setup asterisk with mediant 2000 with avaya [asterisk]-----[mediant 2000]--------[Avaya] when i call from avaya side to ---> asterisk i don't got ringback Sound so how to asterisk genrate tone for calling party is there any soution and what is the problem of

Hi all, Sorry if this has been answered previously, but I have not had any luck trying to find it. I am trying to connect my Asterisk server (1.0 stable, Fedora Core 2, kernel 2.6.8-1.521) to connect to a gateway that can only support H323. I have installed the asterisk-oh323 channel driver (version 0.6.3b) using Open H323 1.13.5 (patched as per asterisk-oh323's instructions) and PWLIB

Hello, I have used the Dial option 'r' before in older Asterisk versions and it behaved as expected, that is, Asterisk would always give ringback audio before the call was answered no matter what. It has been a while that I have used version 1.4.33.1 without any the 'r' option. Now that I had to use it for a while, I noticed that 'r' would not give ANY audio until the

Whoops, email client auto-filled dev previously. Let's try this again. Michael Maier wrote: <snip> > Ok - but this doesn't seem to answer my main question: > > Why must > > progressinband=never > > be applied especially if asterisk uses it by default? The big difference > between w/ and w/o it is: The default in 13 is "no" which still

Hello! An extension registered at asterisk 13.23 initiates an external call (pjsip). After the Invite, the callee (-> ISP) sends a 100 Trying 183 Session Progress (*without* SDP) Asterisk now sends to the extension: 183 Session Progress (*with* SDP) 183 Session Progress (*with* SDP) (really two times) The callee meanwhile sends 180 Ringing (*without* SDP) which is

Hello everybody! I have an Asterisk 1.2.12.1 server with SIP as the VoIP protocol. What I want to do: I want to talk to the "outside world" via MGCP. I suppose I must set an MGCP peer to route outgoing calls. So, I must set the endpoint syntax of the Asterisk server (Asterisk will act as an MGCP gateway and will talk with an MGCP Gatekeeper) and with other MGCP gateways via

The problem I have had with many SMS email gateways, is that they do not allow to reply back. Of course, email is a superior service over SMS, but does not really replace it ( - at least not until all the mobile phones have email access on them...). It would be nice if the SMS service would be available not only on GSM/GPRS/etc., but also on any telephone number in the world - or at least on