similar to: 'Intercom' before call transfer

Hi there, I'm wondering if there is a way to assign a different Caller ID to each Zap interface. I have 3 Digium X100P cards, and I'm sure there must be some way of configuring zapata.conf to allow each line to identify itself with a different Caller ID string. Many thanks, Steve -- Steve Foy | http://www.unite.net UNITE Solutions | Tel: 028 9077 7338

Hi there, I'm having some trouble with getting Asterisk to make a call, I think it should be quite easy, but anyway... Using the following file contents: ## Channel: Zap/3/<TEL NUMBER HERE> MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: phones Extension: 502 Priority: 1 ## Extension 502 is simply one that plays a sound back. When I dump this file into

Hi, I've got a fairly working Asterisk setup, with a few minor glitches, one of which is very very irritating. Sometimes, during a call, the remote end just drops off. We're using software SIP phones (SJPhone) connecting to * then out through analogue lines with X100P cards. There is nothing in the logs and nothing on the console, the call just seems to 'go away'! Can anyone

This strikes me as something that should be really very simple to do, but I can't figure it out. Is there a way of logging all SIP debuging info to a file somewhere? It would help me greatly! Cheers, Steve -- Steve Foy | http://www.unite.net UNITE Solutions | Tel: 028 9077 7338

Hi there... I have a simple * setup with about 11 Soft phones (SJ Phone). The clients don't support a consultive or supervised transfer (I believe that's what it is called). Tris is a feature much desired by the powers that be and they want me to "make it work" :) I was wondering if there was a way to do this with and AGI script or the like so that when Staff 1 gets an external

This message is in response to Flash operator problem. My op_server.pl seems to be same. I also created the variable.txt to the /var/www/html/panel folder and when I run htt://192.168.0.0/panel it just says at the bottom transferring data. I don't see anything on the screen. I also checked my manager.conf file. I was able to telnet into the manager interface and it's running fine. So I am

Has anyone from digium looked at why there is a 30 min to 3 hour lag on messages on this list? I.e looking at the last 50 messages I've received, the lag is about 90 minutes between the time sent and the time received. Sometimes this drops to as little as 4 minutes. Is this problem worse for me because my email address starts with "w" and my copies of the emails get sent after

Does anybody know why I get this error when using zapateller: WARNING[1209214400]: File rtp.c, Line 327 (ast_rtcp_read): RTP Read error: Resource temporarily unavailable It's scrolls until a sound is recived from the line, then it plays the zapateller tones. /Mike

I've seen a lot of traffic on the list recently about which phones can do supervised transfers and which cannot, and I have to admit that I'm a bit puzzled. Our existing PBX, which is software based, handles the transfer functions for our call center -- the agents never touch their phone, and instead use software. We can plug any old phone into it, and it'll work just the same. So

Is it true that every time we make a change in the configuration file we need to restart the asterisk server. This will not be practical in the production environment. Thanks,

Hi, I'm looking for a reliable provider to use mainly for outgoing calls in the UK, incoming isn't so much of a worry as I think I'm going to accept them over ISDN. Cheers! Steve -- Steve Foy steve@narnian.org

I just put in a Audiocodes Mediant 1000, which seems to be working well except for one annoyance. I am using Polycom 501's and 601',s and if I do a supervised transfer of a PSTN call where I complete the transfer before the 3rd party has answered, the PSTN party hears dead air until the call is answered or goes to voicemail. I'm not really sure where to start my troubleshooting. Any

Hi. I was wondering if there's a way to do supervised transfers on the budgetone 102 . Blind works ok, but can't do supervised. I thought that with the flash button that could be possible, but seems I'm wrong .... matteo.

I made the post below earlier today. I'v since removed all NAT from the equation and the problem still persists. Basically I am trying to transfer a call. The transferring phone sends a REFER message to asterisk with a call id that Asterisk doesn't know about. Surely, surely.... someone else must have seen this? hermes*CLI> sip show channels Peer User/ANR Call ID

Hello all, Does anyone know how to successfully transfer a call using the GrandStream Budgetone 100 phones? I've read multiple posts talking about hitting flash, the dialing, then flash again, etc. Some posts talk about using the transfer button, then dial, then flash. Anyways, it seems that I am able to put the caller on hold (whereas they hear hold music) by pressing flash or

Good...it almost works fine! I just have 't' in command Dial, but i also have 'T'. Is it a problem ? This is my Dial() exten => 605,1,Dial(${GIORDANO NAT},60,Ttr) I have only a problem: A and B are speaking, B calls C and ask it if wants speak with A, C accept but if B hang up A is waiting and C get busy tone. To make it works B don't have to hangup but habe to press

I have not been able to find anything that relates to this problem. The agents are using Cisco phones. Calls goes into a queue. but once an agent picks it up it cannot be transferred. However if they call directly to the agents extension it's not a problem transferring calls. It sounds like a misconfiguration but I cannot see what's wrong. Any takers? -- Steve Szmidt "They

Hi all, I have a user who is having trouble transferring calls, using a Grandstream GXP2xxx. Here's the use case that I've seen: I call the user from phone A and he answers on phone B. Then, he hits the transfer button on his phone and dials an extension that is reachable by him, but not by me, based on administrative policy. However, the Asterisk logs indicate that the new call is

Is there a way to keep the incoming CallerID from the PSTN and pass it onto the sip phone receiving the supervised call transfer? The receptionist receives the PSTN callerID, performs a supervised transfer, we get her local SIP callerID, not the original callers. The main reason we would like the true callerID is for asterisk monitor to name the file correctly for call records. Is this

Hi all, I'm quite new on this mailing list, and I discover the asterisk world. I m experimenting a PBX with SIP phones, grandstream budgetone (not expensive for tests) All the features I need work just not one : the supervised call transfers. I know there are a lot of posts about that, but none gave me the correct answer (unless I missed it). Here is my config : 2 sip phones BT102 with