similar to: SIP/ShoreTel REFER support

Hello everybody. I have a problem with an integration between an Asterisk (1.4.24.1) on FreeBSD 7.0 and a Shoretel 7.5 server. To make a very long story short, when someone behind asterisk call an extension behing shoretel everything work as expected. When someone behing the shoretel server call someone behind asterisk the first 10 seconds of the call seems ok but then the line is dropped

I've been anxiously awaiting ShoreTel version 6 because of it expanded use of SIP. My plan was to upgrade out ShoreTel server at our main office to version 6, and use Asterisk in our small remote offices, and have them all be able to directly dial each other's extension. (i.e. CEO in main office can dial ext 401 and get directly to secretary at small remote office, and vice-versa)

Hi, Someone who has had experience with shoretel VoIP systems, can you please give me a run down of how Asterisk is either better or worse? I am completely unfamiliar with Shoretel systems, but someone had suggested we look into them. I said, you bring your Shoretel features, and I'll show you 10 things Asterisk does or can do that Shoretel doesn't do. I still believe that's

We have a Shortel system at out main site. We're putting Asterisk servers at several smaller remote sites. I know I'll be able to get the Asterisk servers to talk to each other via IAX, but can they talk to the Shoretel server? Basically, I'd like to be able to, from the main site with Shoretel, dial an extension, and reach that phone at a remote site, and vice-versa. Thank

I've got a ShoreTel 210 MGCP phone drops calls. My packet capture indicates that the phone may be trying to renew its registration with *, but reports Restart Method of Disconnected (frame 2), then * seems to take that as a sign that it has lost the connection and closes things down. The phone, meanwhile, seems to think it can continue the conversation until a few ICMP "port

Okay, so I'm a noob. Asterisk looks very promising, so I say "thanks" and "good job" to all who contribute. My * test box is up and running with soft phones using IAX and SIP, so now I'm on to testing hard phones. I borrowed a couple ShoreTel 210 phones from somebody who had them on hand but they only support MGCP. I see that there's an mgcp.conf in

[BEGIN] 2018/8/24 16:57:41 [root@localhost puduct]# guestmount -v --rw -a /home/kvmsystem/VSOS_2G.qcow2 -m /dev/sda1 /mnt/ids/ libguestfs: trace: set_verbose true libguestfs: trace: set_verbose = 0 libguestfs: create: flags = 0, handle = 0x7f243deb9920, program = guestmount libguestfs: trace: set_verbose true libguestfs: trace: set_verbose = 0 libguestfs: trace: set_recovery_proc false libguestfs:

Hello! Anyone with some insight into the oh323 channel driver please shed some light on the code block below from wrapendpoint.cxx. When enabling trace on the channel driver i get this, for me, strange debug info: WrapH323EndPoint::OpenAudioChannel: Direction => PLAYER, Buffer => 320 WrapH323EndPoint::OpenAudioChannel: FrameSize 8, FrameTime 8, TimeUnits 8

[I'm sorry to trouble the list with this, but this is the only way I know to contact the person concerned] This message is for Stephen Karrington - it appears that you have over-agressive 'spam' filters and we can no longer email you. Please rectify this if we are to have meaningful conversation! The original message was received from Linus Surguy

We have an Asterisk server that speaks IAX2 to Magrathea to get to the PSTN. Our local phones are a mix of Cisco 7940s and Grandstream BT100s all configured for SIP with silence-suppression disabled. Everything is configured to use a-law encoding. The version is: sip*CLI> show version Asterisk CVS-05/06/04-18:45:57 built by root@sip on a i686 running Linux Incoming callers are complaining of

I have a client who is using a Shoretel PBX. This PBX apparently does not send DTMF information OOB, but instead sends this inband via the B-channel. This is traversing an Asterisk box via a PRI. The user calls the IVR (1-800-CALL-DHL), receives audio, but is not able to present DTMF to engage the IVR. With some light research it appears that the DSP is not activating until the call is

I am looking to install a web interface for Asterisk to transfer calls and look who's on the phone. If anybody has a working web interface please let me know. I installed the www.asternic.com (operator) But when I bring up my web browser it says transferring data and does not bring a browser. -----Original Message----- From: asterisk-users-admin@lists.digium.com

Hi, I have encountered a DTMF issue. My scenario: Access carrier-----sip----> Asterisk-1.4.25.1-----sip---->CiscoGW-----ISDN----->TDM Switch the access carrier sends to Asterisk out of band (rfc2833) dtmf, Asterisk forwards it with SIP INFO method to Cisco gateway, but on TDM switch every digit is duplicated. Is it possible that the carrier sends inband along with rfc2833? Kind

The freenum.org beta continues to roll forward. If you have an Asterisk or SER SIP gateway/proxy, please see if you can make some sense of the examples below and install them in your system. Your users will hopefully be able to dial toll free numbers in various nations just like they dial regular numbers in those same country codes. I'd like to get some additional people trying to make

We're in the process of testing some equipment and configurations and to do this we have setup a UK PSTN Gateway to Free World Dialup. Simply dial 0845 004 5566 (UK local rate call) and at the prompt enter the FWD subscriber number - within a couple of seconds you should be connected. We can also terminate UK 0800/0808 numbers for SIP/IAX -> PSTN calls, at the moment we don't have an

SIP user => Asterisk 1.6 server => SIP Trunk => external destination: works AMI script => Asterisk 1.6 server => SIP Trunk => external destination: Failed to authenticate on INVITE to '"asterisk" <sip:asterisk@(ipaddr)>;tag=alphanumeric' I?ve tried doing things like ?include => contextwithtrunk" in various places, googling, re-reading relevant

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > George Pajari > Sent: Thursday, June 09, 2005 10:19 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Voicemail and MS Exchange Synchronization > > > We have a customer considering

Hey, all. While ADA can still be downloaded, that's about all that I see. No development, no recent mention, and -- perhaps worst of all -- it appears not to work properly under 64-bit systems. So, assuming Digium's abandoned it, are there any suggestions of alternatives? Right now, I'm replacing a Shoretel system, and I'd *dearly* love to avoid the incredibly fat client they

Hello Guys, after an update to cvs head (thanks oej!) my CiscoGW can now flag unkown caller's to Number AND Name "Unkown". Before i again open a new bug (which isn't a bug :-)), can someone confirm this: - PrivacyManager does not recognize this as an unknown number - it's not possible to set ANY CID with SetCallerID, it allways stays on Unknown (with chan_capi i had

Sorry for cross posting, but I somehow came to think that this question might possibly be more suitable in this mailinglist. I am currently fiddeling around with Fedora Core 2. Does anyone of you have any pointers on where to start reading to compile OCFS 1.0.13 on a 2.6.8 kernel? Is it at all possible? Best, //magnus -----Urspr?ngliche Nachricht----- Von: Magnus Lubeck [mailto:ml@inwork.ch]