similar to: Best way ro run 2 or more asterisk servers?

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've been struggling with an ongoing problem the last month. Here is the layout of the wiring: T1 from ISP > DiTech Echo Cancel device > Voice Channel-Bank (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1 server zap card > fax channel bank (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1

Hi, I need some information on DID and Callback. Please read-on: Question on DID (User1 Calling User2 via normal Telephone line and sending its CLI: Connectivity is as below: User1 ==PSTN==> DigiumE1/Asterisk1 ==INTERNET==> DigiumE1/Asterisk2 ==PSTN==> User2 1. Can User1 make a single stage call to User2 via Asterisk1? Currently User1 is able call User2 on Two Stage basis (Asterisk

Hi All, We are experiencing a a problem when running calls over IAX with g.729. The call flow is as follows: Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier The first Asterisk system is running 1.2 and the second is running 1.0. When using g726 from the handset all the way thru to Asterisk2(then 729 for the carrier leg) calls go thru fine, but when using g729, there

Hello, I have the fowling scenario: fxs[asterisk1]-----iax-----[asterisk2]e1----e&m---PSTN I want to know the steps to transmit fax from a machine connected to the fxs to a fax machine on the PSTN. The same for dial-up's. Is it possible only with a/ulaw ? What configs I need in asterisk1? Thanks in advance Eduardo

Hello, I have transcoding card TC400P installed in server running Debian with Asterisk 1.4.23. Everything seams to be fine and after I boot up server I see in dmesg: 7.590966] Zapata Telephony Interface Registered on major 196 [ 7.590966] Zaptel Version: 1.4.12.1 [ 7.590966] Zaptel Echo Canceller: MG2 [ 7.610963] zttranscode: Loaded. [ 7.618969] wctc4xxp: tc400b0: Attached to

Hello, I'm a little confused on DTMF. A sip peer is registered on two Asterisk servers. No dtmfmode is set for them, the sip peer is 999 on Asterisk 1 and 999 on Asterisk 2. They both register on each other. A call comes in on Asterisk server 1, provider 1, dtmf=inband. Then the call is transferred to Asterisk 2: RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/12351 at

Hi All; Did anyone try to implement (installation and configuration and running) for more than one asterisk instance (two or three instances), where each asterisk instance to work on a difference IP than the other where the server already has more than one IP address. We need to implement this situation because in case we need to do testing for any scenario of configuration, then other

Hi!I've a question: I've 2 asterisk, I want pull the ethernet wire and then reconnect it after 5 second, using the VRRP protocol, where must I set the IP for the connection goes on the second asterisk? I want this: I call to asterisk1, then I pull the ethernet wire down, vrrp makes up the other asterisk but not the audio streaming...the callers are always pointed to asterisk1, but for the

Hi I am trying to make this setup work phone1---g729---asterisk1---sip---asterisk2---g729---phone2 I have tried several configurations but none worked I keep getting transcoding errors I have installed one g729 licence on each asterisk, but I can't verifiy because the show g729 command is not available, I use 1.2.17 Do I need 2 g729 licences per asterisk ? Do I need to register

I am getting quite frustrated today, so please bear with me. I just installed Fedora Core 4 (was running RedHat 9 with a working Asterisk) now my Fedora does not appear to be recognizing my X100P (clone) at all. Hardware browser just shows them as unknown device. driver: hisax So, of course, my zaptel drivers do not work and therefore my asterisk does not work. any help would be greatly

Hi all, I need to test the following scenario: +-----------+ +-----------+ | asterisk 1| | asterisk 2| +-----------+ +-----------+ | | | | _______|__________________|___________ | | | | | | +-------+ +-------+ | ATA 1 |

Hello.... I have the follow situatuion: < ISDN > | | V E100P |----------------| IAX2 / g729A |----------------| T100P | Asterisk1 |- - - - - - - - - - - - - - > | Asterisk2 | - - - - - -> |--------------| | | | | | Zhone | ----------------- ----------------- --------------- Here's the situation: I receive calls from the PSTN

This is not really a problem just something I noticed in my testing.. When two or more Asterisk servers are connected by IAX2 trunks it does not make use of any "shortest path" type system.. (maybe this is still planned somwhere down the line, but may come in handy to those who have multi asterisk installations) Here is the setup.. UA1--- Asterisk1----[IAX2 Trunk]---Asterisk2---UA2

Hi all, Is it possible to install more than 1 asterisk in a single server? If yes, what do I need to set and take care? Rgds, ango

hi, i have problem with pass-through faxing with this scenario hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen virtual) - linksys ATA i can fax to fax2mail on hylafax but after upgrade asterisk2 to 1.4 faxing is not working hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.4.X(xen virtual) - linksys ATA configuration is same do you hava any idea what is

Following is my scenario to connect back to back PRI of two asterisk server. PRI cards are Sangoma A102D [Asterisk1]------------[PRI]-Cross Cable---------[Asterisk2] Asterisk1 ; Span 1 (MASTER) switchtype = national ; commonly referred to as NI2 context = from-pstn group = 0,24 echocancel = yes signalling = pri_net channel => 1-23 Asterisk2 ; Span 1 switchtype = national ; commonly

Thanks for that, I am however slightly concerned that due to the fast moving asterisk project (with new versions coming out regularly) that digium may start phasing out support for 2.4 kernel, I would like to settle on an OS for my customers and don't want to have to readdress the situation in one year because the 2.4 kernel is no longer the supported /stable version. Does anybody believe this

We're trying to PQ (Priority Queue) packets on a Cisco using ACL's. What we're trying to avoid is hardcoding the IP address in the ACL. We were trying to match by TOS set by Asterisk however it seems we've run into a snag where the packet TOS tends to get reset somewhere on our network. Has anyone had this issue? We're running Cisco everywhere inbetween (even the switches). Is

hi guys, do you have any suggestions on how to connect 2 TE410P via E1? (for simulation and testing purposes) asterisk1 --> TE410P ----> ? ---------> ? ---->TE410P -->asterisk2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030907/698cd499/attachment.htm

Hi I have the following configuration | UA1 --|------ asterisk1 -----------------------+ UA2 --|------ asterisk2 -----------------------+ DB UA3 --|------ asterisk3 -----------------------+ UA4 --|------ asterisk4 -----------------------+ | All UA is located in the same area. A seperated PC is used as a centralized DB for storing a common dial plan, user account and register