similar to: Problems with Call Forwarding on a 7960

Greetings all.. We have an inbound/outbound PRI installed and terminated on a T400P ? Digium Quad T1 card. We?re seeing an odd problem when sending $CALLERIDNUM when calls from the PRI are forwarded back out to the PSTN over the PRI. The $CALLERIDNUM is not being sent out along with the call. It?s sending the phone number of the PRI itself, rather than the $CALLERIDNUM information. Yes, we can

I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip extensions and a regular phone connected to the box. All routing works fine from the regular phone connected to the box, whether its going to FWD, broadvoice or the PSTN. The problem I am experiencing comes from making calls from the sip phones. They get routed correctly to the sip and iax trunks but when making calls

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

Hello How do I fix this.... [IPComms-in] exten => ${IPCCIDN01},1,Noop(${DATETIME} ${CALLERID}) exten => ${IPCCIDN01},2,SetCallerID(${CALLERID}) exten => ${IPCCIDN02},1,Noop(${DATETIME} ${CALLERID}) exten => ${IPCCIDN02},2,SetCallerID(${CALLERID}) exten => ${IPCCIDN03},1,Noop(${DATETIME} ${CALLERID}) exten => ${IPCCIDN03},2,SetCallerID(${CALLERID}) exten =>

Hi forum, I have been fighting days and days configuring FWD and asterisk with NO success I have the following scenario. My sister in Spain with FWD dialup client My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone. Spain LAN FWD

Got one weird one and one prob easy one. 1. I have upgraded our BT101's to Program--1.0.5.0 Bootloader--1.0.0.17 HTML--1.0.0.34 VOC--1.0.0.6 after doing this i have some phones on different subnet's ie 255.255.255.248 or .192 or .252 and i am now unable to login to these phones from different subnet's . I have one at home which is on a .248 ( Using an external IP for the phone )

On Tue, Jul 18, 2017 at 15:33:24 +0200, Martin Kletzander wrote: > On Tue, Jul 18, 2017 at 06:06:45PM +0800, wyang wrote: > >Hi, guys > > > >I was executing the following command to lively migrate local storage to > >network storage, > > > > [root@192-168-152-51 ~]# virsh blockcopy > > a75056a4-44e0-4b97-bc77-78142ee17038 vda —xml rbd.xml —wait

I'm hoping someone can help me out with this. I am basically just trying to separate my outbound and inbound calls into separate contexts instead of having everything in a single context. Any help would be appreciated. Perhaps I've missed something really obvious.... Here is the network layout: <remote> <--TDM400P--> <nattedbox> <--IAX--> <liveipbox>

I'm trying to connect an asterisk server via oh323 to a Lucent iMerge. I patched the code due so that Lucent can handle asterisk's ver4 h323 http://www.voip-info.org/wiki-Asterisk+Lucent+iMerge+Configuration I can now successfully dial in to our company over multiple lines. The issue is when I dial out The first outgoing call connects to an outside user A The second call drops the first

I am trying to SetCallerID to a variable I have defined. This obviously is wrong. It actually sets the caller ID to $NEWCALLERID. I have search through the examples on wiki but wasn't able to find something similar to see what I was doing wrong. Could someone tell me the correct way to SetCallerID to a defined variable? exten => 2125551212,5,SetCallerID({$NEWCALLERID}) exten =>

Hi, Does anybody know if it is possible to change the tones on a Cisco 7960 ? The default seems to be US standard in BUSY, REORDER, DIALTONE etc etc. I guess there has to be a way to change them ? /Regards Mike

My number is not submitted. I updated my asterisk but this error still occurs coz of the "" in the SetCallerID tag thats why it will be a empty SetCallerID is submitted. Is there a fix to correct this error? -- Executing SetCIDNum("SIP/31-752a", "4989427xxxx") in new stack -- Executing SetCIDName("SIP/31-752a", "4989427xxxx") in new stack

On Tue, Jul 18, 2017 at 06:06:45PM +0800, wyang wrote: >Hi, guys > >I was executing the following command to lively migrate local storage to >network storage, > > [root@192-168-152-51 ~]# virsh blockcopy > a75056a4-44e0-4b97-bc77-78142ee17038 vda —xml rbd.xml —wait —verbose > —pivot > error: argument unsupported: non-file destination not supported yet >

When somebody call me on my pstn # cable connected to my fxo card it does not work when I check my computer the following error shows Connected to Asterisk CVS-v1-0-12/05/04-19:46:25 currently running on asterisk1 (pid = 2160) Verbosity is atleast 3 -- Remote UNIX connection -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at incoming,s,1 failed so falling

Hello, I've been racking my brain over this for much of the day so I thought the list would probably be more helpful. A few days ago I upgraded from Asterisk 1.2 to Asterisk 1.4.5. Everything appeared to be working properly. However, on the first business day, we realized that when transferring calls (not using call parking, using the built in transfer buttons on a Cisco 7960) would not

Hey guys. Thanks for the help on the Pattern matching, I got that working pretty nicely. the next problem I have is that I'm using an include file, but its not really working... In my extensions.conf: [incoming] exten => _NXXNXXXXXX,1,SetCallerID("Unknown Called Number") #include "numbers.conf" exten => _NXXNXXXXXX,3,Macro(Number,1000,${EXTEN}) [macro-Brand]

Asterisk 1.2.16 I have an extension "102" with a Polycom 430 I am trying to protect against forwarding loops If I set the phone to forward the line to itself, extension 102 I get the following -- Got SIP response 302 "Moved Temporarily" back from 206.83.240.18 -- Now forwarding Local/102@mycontext-b2ee,2 to 'Local/102@mycontext' (thanks to

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Robert Withrow > Sent: Tuesday, June 15, 2004 12:32 PM > To: Asterisk-users > Subject: RE: [Asterisk-Users] making * more like a normal pbx (ciscoata- > 186) > > On Mon, 2004-06-14 at 19:34, Reid A. Forrest wrote: > > I've

What the heck does this mean? This is the first time I've seen this. Calls were going through ok for a couple weeks now. Dec 6 09:22:24 WARNING[1121866688]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/1400-45fb(4) to IAX2/simple/2(256) Dec 6 09:22:24 WARNING[1121866688]: app_dial.c:998 dial_exec: Had to drop call because I couldn't make SIP/1400-45fb

Hello all this morning i followed wiki in subject to replicate my active directory, but it fails with this error: [root at dc1 etc]#  samba-tool drs showrepl Default-First-Site-Name\DC1 DSA Options: 0x00000001 DSA object GUID: 8ba457e4-815d-4bd3-a748-8b5ddb53fd5f DSA invocationId: 834770f4-c5a7-48c7-bc77-66e2cf37e557 ==== INBOUND NEIGHBORS ==== DC=ForestDnsZones,DC=lxcerruti,DC=com