similar to: SIP response 482 "Loop Detected"

Dear All : I need to mask the caller ID and pretend to make a transfer call from another extension : exten => 558,1,Answer exten => 558,2,Playback(soundclip) exten => 558,3,Dial(SIP/472@callman) The scenario is like this : Someone is calling 558 at my company - he will hear a soundclip voice message then I will direct it to extension 472 I need 472 to not see the extension of

Dear All : I need to link my HQ to some Remote Sites - I need a Router which supports VOIP , and VPN Also the Router Should has 3 FXS ports and 1 FXO ... The call should be routed from the Remote Site to the HQ through a VPN tunnel ( 3DES ) ... Any Advise ? Mohamed Farid ,, Notice: This e-mail (including attachments) is confidential and is intended solely for the addressee. Unless you are

I did create a Voice Mail Boxes for Some Cisco Phone Sets ,, I can now record a Voice Mails , and I can hear them ,, but I am not able to configure the Voice Mail Button on the Phone Sets to directly listen to the Voice Messages .. Also when there is a message the Phone doesn't light to give a notice to the User of the phone .. Any solution ? Eng. Mohamed Farid ,, Mediterranean Smart Cards

All, Just came across the prompt #3 from inside the top menu of VM in latest stable. Allison does not announce the prompt, but if you know it is there, you can press 3 & successfully follow the prompts from there to send your message to other users on the system. But, of course, obviously, I am asking: how do I resolve the situation whereby the users are not hearing this prompt? (since most

Hello Before I order a Travla C156 case (http://206.14.132.88/products/Travla/c156/C156.html), a Via mini-ITX motherboard (either the fanless ME6000 http://idotpc.com/TheStore/pc/viewPrd.asp?idcategory=50&idproduct=4 or the fan-equipped M10000 http://idotpc.com/TheStore/pc/viewPrd.asp?idcategory=50&idproduct=163 ) , and a PCI FXO card from Digium or OpenVox... has someone already

Dear All : I have a problem with my MeetMe ,, I need to add the latest update of MeetMe Software so that I can add an announce when entering/exiting the Conference Room .. I saw a lot of WebPages are talking about 'i' option. How can I get the latest update ? and how can I recompile my Asterisk to use the new update ?? Thanks ,,, Eng. Mohamed Farid ,, Mediterranean Smart Cards Company

I'm trying to join a Samba 3.0.4 (compiled from source on Debian) to an Active Directory as a member server. I believe Kerberos is configured correctly as kinit creates a ticket for the realm. Executables appear to have support for Kerberos and LDAP (smbd -b | grep KRB and grep LDAP) return OK. When I try to join the AD with net ads join -U myadminusername I'm prompted for my

Hello everybody, im newbie in VoIP, but find this project asterisk very interesting, i tried to install and its a great sw, i really get sorprised about all of its functions, we need to use the asterisk server in conjunction with cisco callmanager. We have a Cisco Callmanager 4.1 and the clients are softphones from cisco IPCommunicator, but all the support service of our company are linux

Hi, ? I am currently trying to get a Cisco Callmanager 4.1 and an Asterisk server (1.6.2.21) to talk via a SIP trunk so I can use the Voicemail component of the Asterisk (all the phones are associated with the Callmanager). The connection seem to be there. When I do a "sip show peers" on the Asterisk server?I see the Callmanager as Monitored and online however I can't get any calls

Off topic, but involves an Asterisk deployment in a roundabout way. Anyone here intimately familiar with Cisco Callmanager (Version 4-5), that can tell me where a directory of the standard system voice prompts for Callmanager might be obtained? I am looking for the text and filenames of the standard prompt set that ships with Callmanager, have been all over the Cisco site and I can't find it.

Hi folks, I know this isn't an Asterisk question, but I'm really desperate and wondering if someone could help me. I apologise for the off-topic post. Cisco phones connected to CallManager can forward calls. But when they do, CallManager conserves the originating caller's ANI in the new leg that is built. I cannot find a way to get it to rewrite the ANI to be that of the phone.

Hello We have integrated cisco callmanager 4.1 with asterisk and we can dial from cisco to asterisk but we're getting an error if we call from asterisk to callmanager. This is the error I'm getting anybody can help me? Verbosity is at least 3 -- Executing Dial("SIP/2234-e084", "SIP/cme-pbx/4455") in new stack -- Called cme-pbx/4455 -- SIP/cme-pbx-25ae is

Does anyone have the .bin file(s) to convert a Cisco 7940/7960 back to the CallManager image? I want to start playing around with the chan_skinny addition, but it seems the .exe's from cisco want to open a connection to a SQL server or CallManager (which I don't have).

I have installed Asterisk@Home on a PC here and need to have it forward calls to the PSTN. We have Cisco CallManager 3.3.4. However I found out that this version doesn't support configuring SIP Trunks. Is there an alternative solution. Thanks Walid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Siggi, > > 7960 and then "Call Ended" on the Display (curious about that !!!). > > That seems to be normal for the 7920. I've sniffed the registration > procedure with Cisco's newest 3.3(3) CallManager (+patches), and it's > doing the same thing. Maybe that's some odd way of testing if the > CallManager ("CCM") really works... >

I am new to Asterisk and currently work mainly with Cisco Callmanager. With Callmanager I can setup partitions and call search spaces to determine where a given phone can and can't dial. Does Asterisk offer this type of functionality, and if so how? Blake Parker -------------- next part -------------- An HTML attachment was scrubbed... URL:

Callmanager does nothing than construct and tear down calls and the actual RTP stream does not flow through the Callmanager but is direct from IP device to IP device. How does this work with Asterisk? I read something that lead me to believe that Asterisk has to process the entire call, is this the case? Blake Parker CCNA Network Engineer Alacare Home Health & Hospice, Inc. Email:

Hi..... I have just successfully setup Asterisk with 2 Cisco 7940 phones (converted for SIP) and a SIP softphone on a W2K box.....and it all seems to work very well.....to those who wrote this software, it is really cool. Anyway, I am new to this software, and I have a lot of questions which I am hoping someone on the mailing list might be able to answer for me.....I am basically trying to

Hi, Where can I find information on H.323 for Asterisk and/or integration with Cisco CallManager in particular? <http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration> I have oh323 working on Asterisk. Since the CallManger I am working with is running 3.3.3 I cannot use SIP... Thanks, Adi