similar to: Using channel banks

I have the following setup: AnalogPhone1--TDM400P-ASTERISK---via SIP---Softswitch--------POTS Phone2 When I call from AnalogPhone1 to Phone2 I hear a ringing tone and all is well. When making a call from Phone2, I get a dial tone but after dialing the number I hear nothing (no ringing tone). On Asterisk console it says that a call is coming in and that it is ringing Zap/2. I can also hear the

I have following setup: AnalogPhone_1--TDM400P--Asterisk---SIP---[Softswitch]----POTS-AnalogPhone_2 I can call from AnalogPhone_1 to AnalogPhone_2 and all is fine. When I call to AnalogPhone_1 from AnalogPhone_2, AnalogPhone_1 rings BUT I hear no ringing tone AND when someone picks up AnalogPhone_1, there is no "sound" and parties on both end cannot hear each other. Seems that no

Greetings... We've been having some interoperability issues between Asterisk and an AudioCodes Mediant 2000, and, well, I gotta use the Mediant 2000 somewhere. So, I've been pondering using iptel.org's SIP server (SIP Express Router) as a "front end" for PSTN calls going out to the Mediant, while using Asterisk for everything else. Has anyone done something similar, or

I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway working with asterisk. The Idea is the following. PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} -- Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic system has FX0

Hi everybody, I want to read to debug messages and try to interpret them but they happen too fast, how can I log these guys to a file, or is there a file like this already? I checked the /var/log/asterisk but there isn't much interesting there yet? How can i turn on logging for SIP,IAX and other things? Thanks, Umut

Hi I am trying to call from Linphone on extention 109 to Xlite on extention 108 and I get this error ---------------------- to 216.75.167.18:5068 WARNING[98315]: File pbx.c, Line 1133 (pbx_extension_helper): No application 'Dial ' for extension (sip, 108, 1) == Spawn extension (sip, 108, 1) exited non-zero on 'SIP/sergeXlite-be43' --------------------- Can you tell me what

I was wondering what are the values for sending dmtf via sip info. I mean, when I use dtmf relay via sip info, the sip/sdp message contains a Signal=X where X is the dmtf. That's ok for dtmf 0-9 . but what when dtmf is * or # ? we must send signal=# ? I ask that because I noticed that budgetones phone sends out * as signal=10 and # as signal=11 . but asterisk don't detect them, 'cause

Hi all! I am looking for some free software to monitoring all the calls that are being done in my network. Which telephone are connected, how long are the calls, quality of service, bandwidht,etc. If someone knows about a good one, plesea tell me. Regards, Mireia

Hey all...I'm just getting my * setup and right now all I have is an FXO but no FXS. I wan't to get rid of telemarketers by having * pick up the phone if there is no CID present, give the caller the Zapateller tones and then ask the user to input their phone number via Privacy Manager (yes I realize that this won't get us any where given that I can't re-ring the phones without FXS

Here's a clip of comments lifted from a Cisco bug list. This will be perhaps useful to those of you who have just purchased a Cisco phone off eBay. JT ------------- (1) Short problem description: Documentation on how to load SIP image on phone with skinny software (2) Longer problem description (what happens): If the phone is loaded with the Cisco Skinny code, then there is a small

I have a problem that sometimes lines will go into what I call never never land. The Asterisk system will put a line with <Zombi> on it when you type show channels it will make the analog phone line dead. And on the CLI it says: astsvr*CLI>Zap/1-2<ZOMBIE>(macro-twoline-exten,s,1)Up Dial Zap/1-2|20|r I have tried to release it with soft hangup Zap/1 & also soft hangup

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?

In the 'me too' vein, I also have an untested 3624 here on the shelf and am waiting on a shipment of T100 cards to play with. Documentation is very hard to come by. Alcatel are certainly the owners of the Mainstreet product line but, without a support contract, any documentation they may have is essentially unavailable as their per-incident fees for support cost more than most of the

1st. - I'm from Poland, we don't have (yet, and hopefully forever) software patents. Is there any free g.729.1 implementation for asterisk? I want to use it for my private use (dialing into inet->PSTN gateway), and I don't want (now) to buy codec, as I don't know if I will be using this service in future (now I just want to test it). Any solutions? Maybe even

I'm looking to buy a channel bank. I'll be using this to basically dual a Toshiba PBX and Asterisk so that we can deploy Asterisk in small steps and have the Toshiba as a fallback in case something goes wrong. What I need is a channel bank that will allow me to take in 12 lines using the FXO ports in the CB and then output those lines to the Digium T100P. Has anyone had experience

When I get a SIP call, I get the following error: *CLI> NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is 'multipart/mixed;boundary="unique-boundary-1"', not 'application/sdp' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, 5147771111, 1) == Spawn extension (incoming,

Hi, Is this card the same as the T410P, after all, it's made by Digium. There's one prior reference on the mailint list[1] but it didn't answer the question. There was also an SS7 status report[2] last June but it's doesn't seem to have lead anywhere either. There was post saying an SS7 release was immenent last September[3], but then silence. Any info anyone would like to

Hi, Using MeetMee(1234,M) on asterisk 1.6.2.9 (debian/testing) we have terrible sound: the MOH is unrecognizable and speakers can't be understood; it sounds "ghostly". However the prompts ("your are the only one in this conference, etc.") sound fine. Our server has a Digium T410P card with two E1 lines going in and the wct4xxp dahdi module. Any idea?

Hi I just came accross this Newbridge Mainstreet 3624 but the Alctel site appears to have zip for reference/user manuals Anyone by chance have 1 of these or a url for the docs ? thx

I did not get any answers to my earlier posting. I hope I have better luck this time- The question: Is it possible to use Asterisk as media gateway controller? I know * supports MGCP, but does that also imply that I can use * to control any third party media gateway (such as one providing media conversion from E1 to AAL2). - DL