similar to: * as a softswitch for pri interfaces

Did anybody figure out how to make dial detect a busy on a zaptel channel on a pri interface when using overlap dialing? According to the documentation dial should return to priority n+101, if the called party is found to be busy. I can see a DISCONNECT message with "user busy" coming from the network when I turn on pri debugging, but the dial application does not seem to notice.

Hi, I am lost trying to figure out how to enable overlap dialing for calls coming in and coing out on a pri span. DISA looked promising at first, but does not seem to support overlap dialing. Just picking up a call by and trying to dial out does not seem the way to do it either. I tried: [dialincontext] exten => 12341234,1,Goto(dialoutcontext,s,1) [dialoutcontext] exten => s,1,Wait,1

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

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?

Did anybody ever come across an affordable, residential cable/dsl router with support for QoS? The ones I've seen so far (Netgear, D-Link and W-Linx) do not seem to support it. I noticed that even email can damage a G.711 stream on an 128kbit uplink, leave alone file-sharing applications. I understand this is strictly related to *, but nevertheless of interest to many of us. Thilo

[ZapRAS triggering a kernel panic] >> Kernel panic: Aiee, killing interrupt handler! >> In interrupt handler - not syncing >> HDLC Receiver overrun on Channel Tor2/0/2/25 (master: Tor2/0/2/25) > > Hrm, I haven't seen this before. Please contact me off-list and I'll > give you more debugging instructions that may be helpful, as well as > enquire additionally

I see in the changelog that ENUM support has been added.. anyone know what this is? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

An application I am running provides a dial tone to my users, read 9 digits, checks whether or not the called party number should be allowed and then dials out using overlap dialing on a pri channel. I.e. exten => _XXXXXXXXX,1,AGI(pm-check-destination.agi) exten => _XXXXXXXXX,2,Dial,Zap/g1/BYEXTENSION|60|CH The AGI-Skript takes about 0.3 to 0.5 seconds (it does a number of rather complex

When I was trying to run mutiple E1s over TDMoE, the zaptel would drivers complain about too little memory, whenever I added more than 31 channels. Requesting 62 channels in a dynamic span gave me ... span creation failed: Cannot allocate memory upon loading the zaptel drivers. How would you go about running, 8 or 16 say, E1s over TDMoE? Would you create multiple dynamic spans or just one large

Hello everybody, Im new to the list and also new to asterisk, Im wondering if I could set up asterisk as a softswitch, I guess for what I've been reading that It could be possible but almost all the info and documentation Ive found so far is about asterisk as a PBX, etc. Im willing to set a small voip wholesale traffic bussiness and Im not quite sure asterisk is the right chocie for that.

>I spoke with someone today who is interested in an IP Centrex solution that >starts with about 3500 extensions in a multi-tenant application. And >growing from there. > >I'm wondering about scalability of Asterisk. I'm trying to put my head >around how to put the whole thing together, if it can be put together. > >The nice thing about it is that if I can show

Hi all, i have configured incoming voip traffic as follows: [voipin] exten => _X.,1,SetCallerID(033283077734) exten => _X.,2,Dial,Zap/g4/${EXTEN} exten => _X.,3,Hangup If the call going out the pri dials with an additional '0' before the dialed number. This is for caller number AND called number. But i can't see anything that says set a '0' more in front of the

Hi Dan, i did it wrong, sorry: curl -v -H "Content-Type: application/json" -u asterisk:asterisk -X POST " http://localhost:8088/ari/channels/newChannelId" <http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world> --data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" ,

Hi all, we are working on a A to B basic Call scenario with early media. On that scenario we get a call from a PJSIP endpoint and we place a new call using ARI. On the created channel we receive a 183 Session progress where we have an announcement regarding e.g. the cost of the call (it's important for us to have this announcement to inform our customers about the costs). Using asterisk

Has anyone setup SIP services with ss7 and lis trunks? If so .. what was used hardware and software.. we're trying to do a SIP -> pstn setup and not having much luck as QWEST keeps pushing dates off (aka trying to screw us over) for our pri lines due to the recent court and fcc activity in regards to unbundled switching and I'm looking for solutions/ideas involving SS7..

I have a number of local users who can dial out on a pri channel using the fantastic new overlap dialing feature. I would like to add a speed dialing feature, such as 1. User picks up and dials out (dial startet with option 'H') 2. User hangs up call with '*' 3. Dialed number is stored in a variable 4. User dials a two-digit extension followed by the # sign to save the stored

Hi All, I've been experimenting during this weekend with asterisk as a softswitch, talk about me being completely lifeless, but let not talk about that. I've been conducting some really funny tests, trying to get an optimal SoftSwitch functionality. Here is my current setup: Source: Windows XP Pro + SJphone Box 1: Asterisk running in PassThorugh mode Box 2: Asterisk running in

Hi Dan, i would do something like this (it is not a copy of what we are doing but an example of how i would do it) Important here is the "--data" and "-H" Option as well as the "variables" section within the Body. I added the callerid function here as well as it is the sample in the asterisk wiki. curl -v -H "Content-Type: application/json" -u

Hello, i've googled and can't find a definite answer, so here goes: I have purchased the Digium TE100P, and am setting up the connection, however the telco i'm supposed to work with does not support PRI/ISDN E1 connections. They only support E1/R2 lines. Is there a way i can make the TE100P work with this? I've not seen any zaptel.conf that supports this. Any workarounds?

Hi, is there a way to enable dtmf detection on zap channels? I am trying to pickup, play a ringtone and the dial out. I.e. exten => s,1,Wait,1 exten => s,1,Answer exten => s,2,Playtones(dial) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 exten => _X,1,StopPlaytones exten => _X,2,Dial,Zap/g8/BYEXTENSION|10