similar to: Unable to obtain dialed number through ZAP

Hi all, I am using Digium PCI board to receive PSTN call through regular phone line. It is no problem for me to receive calls, but I am not able to obtain the Caller ID if the calls are from the phone line. exten => s,1,Answer() exten => s, n, Verbose(1|incoming number is ${CHANNEL} calling to ${EXTEN} routing to ${phonenum} ) exten => s,n, Verbose(1|callid is ${CALLID(num)}) exten

Hi, I have read about SS7 recently and learnt that it is a signalling protocol used in PSTN for call management, setup, etc. The thing that I don't understand is how SS7 plays a role in VOIP. When I make calls between landline and Asterisk via PSTN, I don't need to do anything with SS7. Is it because the SS7 signalling is already done by Asterisk already? From the prespective of

Hi I am working on deploying voip for my company and would like to seek some advice on the number of E1 lines we need to rent. Our telco told us that there can be at most 30 concurrent channels on an E1 line. Typically, what is the maximum number of DIDs that we can allocate to that E1 line before users get frequent "all lines are busy"? We are running a support center with mostly

Hi all, I am using Digium PCI board to receive PSTN call through regular phone line. It is no problem for me to receive calls, but I am not able to capture the destination number through the ZAP channel exten => s, n, Verbose(1|destination to ${EXTEN} ) ${EXTEN} returns 's' instead of the actual destination number. Since I have multiple phone numbers, I want to be able to route

Hi, I am requesting for a E1 connection from my telco. They are asking if I want DSS1 or SS7, and I am stuck here. Could someone tell me the difference between the two? How should I decide which one to use? Thanks in advance for your help. Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

Ok, I'm about to take the plunge, and am trying to decide between Channelized T1 E&M and PRI. I'm getting an "Integrated T1" which will have data and voice capability, all plugged directly into my digium single T1 card. In either case the data piece looks pretty straighforward, just setup the channel properly, hand it off to the linux hdlc layer, and route away.... the

I'm hoping someone can give me a little dialplan assistance. Here is my scenario... I currently have an AT&T T1 connected to a Nortel Optn 11. I recently purchased a Rhino system with a Rhino dual T1 card. What I want to do is insert the Rhino box between the CO and the Nortel on the T1 so I can start migrating users over to the Asterisk system in the near future. But, in the meantime, I

I am looking at buying either a TDM411B or TDM431B to play around with Asterisk at home. What I am wondering is if I could use the TDM431B to divide our phone lines into 3 groups to use as an intercom, but still have them work normally for answering calls and calling outside. I found commands for checking if a channel is in use, so I should be able to do different things depending on whether or

FreeCNAM.org is providing a free CNAM API for Open Source PBX users. This API queries a private CNAM database, and returns standard 15-Character CNAM results. Any entry not already in the database will be queued for investigation, and added to the database as soon as information is located. This system has access to several CNAM backends, and is not a party to any use-limiting or no-caching

Hi, Is it possible to connect Asterisk with a mobile base station to handle call switching? What kind of protocol will I need to use to convert to sip? Any pointer or info will be greatly appreciated. Best Regards, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081119/e74ef6b1/attachment.htm

Hi Folks, having an issue with outbound calls through a VOIP provider. Calls get sent out with the CallerID(number), but where does callerID(name) come from? Apparently not from provider, as we are seeing different (sometime missing) names on inbound calls, different than what we have configured. Apparently this comes from some telco database somewhere? Numbers were ported from a wired-telco.

Hi, I have a Linksys SPA 3102 using as ATA. The call routing is : Phone -> PSTN -> SPA 3102 -> SIP Proxy -> Asterisk The problem I am having is that when the phone hangs up, SPA 3102 can't detect it and relay the CANCEL message. Is this problem with my SPA 3102 config or it just works like that by default? Thanks in advance for your help. Regards, Mark -------------- next

Hi Folks, I'm trying to send CallerID Name information out to the PSTN via a PRI with Bell Canada with no success. With inbound calls (originating from the PSTN) CNAM is received successfully, and we've not had any similar problems with other Telco PRIs, so I'm stumped.

Hi, Does anyone know of any projects that would allow you to use Redis in place of AstDB? By in place of I don't mean for what Asterisk needs but to store values. For instance for CNAM currently we need to use an AGI to connect to redis to pull CNAM. So in place of: Set(CALLERID(name)=${DB(CNAM/${CALLERID(num)})} it would be done with redis for example:

Hi, I am looking for a very low cost way of receiving and sending T38 fax reliably. Is there any possible solution using Asterisk as the PSTN SIP gateay and Digium E1/T1 card? Is there other open source package that can help to accomplish this purpose? Regards, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > brett-asterisk@worldcall.net > Sent: Tuesday, November 30, 2004 2:53 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] National (US) callerid name > resolution for yourasterisk box >

I'm having trouble getting callerid name to show up on my phones (Cisco 7960 and a few softphones) When I look in the CDR database I see the name but not on any phone when being called. I'm running Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC Any help would be great !

I have officially engaged in a pissing contest with the local Telco over PRI calling name delivery. The telco publishes their calling name delivery over PRI feature as being bellcore gr-1367-core compliant. The gr-1367-core spec states that the calling name is to be included as a facility IE in the setup message, or sent in a subsequent facility IE message with an indicator in the setup message

Hey Guys, I'm curious if there's an interest in a free, CallerID database? For those of you in the same spot we are, our current provider only provides us with the CND, excluding CNAM. Would creating a public database, managed by users be worthwhile to anyone? Thanks - Any input is greatly appreciated. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office:

Dear all, I am working on customizing hylafax's faxrcvd script into PHP. Does anyone has any sample or guideline that can share with me to give me a quick start? Two questions I have are: 1. How to simulate the receival of fax without actually sending one? 2. Where can I find the log that is "echo" from faxrcvd? 3. How to I config Hylafax so that it uses my PHP script instead