similar to: No CallerID Transfer Problem

This possibly isn't 100% asterisk related, but I'd like some opinions/feedback... A customer wanted different ring-tones to differentiate external and internal calls. No biggie once I'd worked out that details - they have 100% GXP2000 phones, so adding in the relevant SIP header and altering the phones to suit seems like it's going to be a solution... But I started to look at

Hello all, I think I have most of my AAH 1.3 setup running (Asterisk 1.0.9), but somehow something is not quite right with my vMail setup. I would have sworn this was all working, but maybe I was just dreaming. Anyway here is what is happening, say I am on extension 200 and I want to call to extension 201. If extension 201 is no connected, then it rolls right into vMail with the message the

Hello, when a call comes in and is answered by colleague A, this colleague A sees the CallerID of the external calling number. When colleague A transfers the call to colleague B, attended or unattended, then colleague B sees the number of colleague A on his screen while talking to the external calling number. I expect here that colleague B would see the external calling number on the screen

Hi All, I have registered "abhijit" for SIP in asterisk Server. I am able to register my softphone (SJPhone) to the server using the name "abhijit". But whenever I try to make any calls I am gettinh the following error message:- *CLI> -- Registered SIP 'abhijit' at 172.20.28.85 port 5060 expires 120 May 26 07:34:52 NOTICE[2761]: pbx.c:1738 pbx_extension_helper:

Thanks everyone for the input. After pricing everything we need out, it's not worth trying to get our old system to work, so I've pitched ditching everything and starting over. I'm very excited and hoping they'll go for it. Regardless, I'm going to throw a box together for my house, we have no home phone (just cell phones) so this'll be a great way of testing. All

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm

Hi, I'm using Asterisk to bridge the incoming call to another destination using the Dial command. However, when an anonymous call comes in then privacy information is not passed into the B Leg. For instance, the Privacy header and P-Asserted-Identity aren't copied to the B Leg. Is there an option to give to the Dial command, or another variable to set, to make Asterisk copy such

Hi, I am running a 5 seater inbound call center on 1.0.9-BRIstuffed-0.2.0-RC8h and it's running well. I am now trying to upgrade it to 1.2.4. So I installed 1.2.4 from source and copied all config files from original to the new server. But when a caller lands inside the queue no queue message is getting played. The gsm files are present in proper locations, whcih I am able to play using

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Hello, using Asterisk 1.8.12.2 case : I call with my cellphone to our public telephone number Our receptionist answers the incoming call and does an attended transfer to my colleague ( A ) Colleague answers and the receptionist tells him that I am on the other side. Receptionist transfers the call and I am connected to my colleague ( B ) My question is about the CallerID that the

We have an inbound PRI connected to our Cisco 3825 router which is then passing the calls to Asterisk as SIP calls. We're getting the CallerID number but not the CallerID name. We are seeing the name in the RPID field with a SIP trace on the Asterisk box but don't understand why it's not registering as the CallerID name. Here is a link to pastebin with the Sip trace. In it you

Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

Hi List, Another dumb conversion question (I hope). I installed 10.0 and copied my 1.4 configuration files over. With a few tweaks everything works great except for 1 feature that I specifically went to 10.0 for. When I do an attended transfer, I still get the receptionists caller ID on the transferred phone instead of the incoming callerID. My assumption is that there is some

Hi All I have one issue regarding override callerid when i have anonymous call. I have added PAI in sip header and also set sendrpid = yes in sip.conf but the callerid is not overriding while i am sending call to three digit calling like 911. please give some idea and help for this issue! I am using asterisk 1.4 branch. thanks in advance!! Thanks, Max Alex Voip Developer -------------- next part

Hello all, I am using the Snom 370 phone with firmware Snom370-SIP 7.0.17* *connected to an asterisk 1.2.14 and I can't record any calls using the "Recording" button on this phone. The extension I configured on this phone has the values "Recording on demand", an the voicemail enabled. I am using FreePBX to manage my PBX. How should I configure the "Function

Hello, I configured both asterisk and grandstream 2000 accourding to howtos on the web.. And everything seems working fin. But if i reload asterisk grandstream stops working with BLF. I need to restart the phone to enable BLF again. Any clues??

Hi, I've run into a minor snag trying to pass on CALLERID presence from one Asterisk to another via SIP (both running 13.16.0) I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP. PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has its own callerid values and presence. I pass on those calls to PBX_B via SI, and I'm trying to pass on this

We're using Grandstream GXP-2000 with programmed buttons to the first 5 parking lot extensions. When a call is parked, whichever parking lot extension it's parked on lights up red. We've never used the "announce" part and I'm wondering if there's an option I can't seem to find to disable the announce so the transfer happens faster. Thanks for any help, Ken

We're deploying an asterisk-based phone system at all of our branch offices in an effort to eliminate long-distance costs incurred from the constant branch to branch calls. We're using the Snom 300's at all offices for the desk phones and X100P cards to interface to 2 analog lines. I'm having a problem tuning all the echo out of the system. So far two branches are using the

We're moving a somewhat complicated call center over to an Asterisk system, and I'm looking for documentation on queue/agent configuration. So far I haven't found anything on the Digium or Asterisk websites, and I was hoping that someone could point me in the right direction. Thanks, Devon