similar to: How to Answer QUEUE call through AMI

Hi Friend, Can't get hostname environment variable on asterisk dialplan. Help me about how to get hostname environment variable on asterisk dialplan. I have written "export HOSTNAME" in /root/.bash_profile and when i execute "echo $HOSTNAME" then get right hostname but not success through asterisk dialplan. Get environment variable path right value through below

Hi Friends, need to help. *I have problem about sip : SIP/2.0 401 Unauthorized* Is it require to nathelper module in kamailio ? *what can i write kamailio.cfg file when kamailio and Asterisk on same network?* Scenario is like as : ----------------------------- 1) kamailio server on 172.18.100.74 kamailio.cfg ( nathelpler module ) ----------------- loadmodule "nathelper.so"

Hi all, I want to upgared from asterisk-1.6.2.6 version to asterisk-1.8.0 version. When i execute "make" command for compilation i have seen below errors. In file included from /usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/cdr.h:31 /usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/data.h:233: error: field ?AST_DATA_IPADDR? has incomplete type

Hi Friends, i have installed and configure asterisk-1.8.0. When i have tried asterisk start get below errors and not able to start asterisk. *FD 32767 exceeds the maximum size of ast_fdset!* Thanks in advance. -- Best Regards, Rajnikant Vanza -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Friends, LOCAL/*89/9875784578 I want to match above dialstring into dialplan context. How can i match dialplan extension pattern matching for "*89/9875784578" with including '/' character. Thanks in advance. -- Best Regards, Rajnikant Vanza -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf, but its not yet production ready. Can someone please pitch in about HA feature in Asterisk ? (HA -> High Availability.) Also, What would be the pros and cons of using AsteriskNow over Asterisk ? Are the versions same in Asterisk and AsteriskNow ? We have been evaluating Asterisk for our Voice Application and it seems

Hi all, I need 2 CDR Records of below 2 numbers for outgoing calls, detail is given as below: *96XXXXXXXX88XXXXXXXX* *=> Call file : outbound call generate through below file* Test.call ====== Channel: local/s at outgoing/n WaitTime: 45 Context: outgoing_ivrs Extension: s Priority: 1 Set: contact_no=96XXXXXXXX extensions.conf ============ [outgoing] exten => s,1,NoOP(----- First LEG

I have this in sip show history for a particular channel marked as dead (should be removed) in sip show channels: 1. TxReqRel INVITE / 102 INVITE 2. Rx SIP/2.0 / 102 INVITE 3. CancelDestroy 4. Rx SIP/2.0 / 102 INVITE 5. CancelDestroy 6. Unhold SIP/2.0 7. Rx SIP/2.0 / 102 INVITE 8. CancelDestroy 9. Unhold SIP/2.0 10. Rx

I got a very strange problem with call-hold function. For calls that come in from PSTN and route to a SIP extension. If I put the call on hold, I cannot unhold the call after. The caller would be left with hold music forever. A warning message would be shown on the console usually a few seconds after putting the call on hold: WARNING[17428]: chan_sip.c:686 retrans_pkt: Maximum retries

Hello, I think I met a case similar to the one solved by [1] . Quoting this case : * res_pjsip: Handle deferred SDP hold/unhold properly. Some SIP devices indicate hold/unhold using deferred SDP reinvites. In other words, they provide no SDP in the reinvite. A typical transaction that starts hold might look something like this: * Device sends reinvite with no SDP * Asterisk

Hi, I want to write API for doing some actions. I want to write function for hold special call via AMI.But I can not find any action for this purpose. Is there any action for holding special channel? Regards, Mahdieh Saeed -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everybody and merry xmas. I have a problem with sip phones calling each other inside the same network (no nat, no firewall). You can make and receive calls and pick them up, but you cannot hear anything on any side of the call. But if you press hold/unhold or you transfer the call, then everything works as expected. Ths SIP phones I've tried are Swissvoice IP10s and kphone, it

Through the AMI how can I tell if a call is on hold or not? I am using 1.4.X Thanks, jerry

I'm trying to capture when a call is placed on and removed from being on hold through the AMI in Asterisk 12.3. In previous versions, the Hold event contained a 'Status' field which indicated if the call was going 'On' or 'Off' hold. However, in 12 not only am I not seeing the Status field, but I am not seeing any AMI Hold event that corresponds to removing the call

Using X-Lite FWD soft phone, I can register, get to the 'demo' menu extension, but that's it. Audio starts, then after a few seconds stops, with packets still being passed. Anyoen have any clues? Yes there are firewalls between here and there, yes there is NAT at my end...What ports need punching, is rfc2833 the correct settign or should I use inband or info? TIA, I just

i need to catch the call hold event from my asterisk-java program. Im using net.sf.asterisk.*; for communicating with asterisk server. I need to get the call hold status on my java program . I can able to get the music on hold status but i cannot able to get the call hold status. The events like 1. HoldEvent , 2.HoldedcallEvent 3. UnHold event are not getting fired when the call hold is

Dear All I want to use asterisk for the following Senario and Need help to find a SAMPLE extension.conf Incoming call >>>>>>>>>>>Asterisk >>>>>>>>>>>>>>GSM Termination Gw first leg second leg What I want to do is putting first call leg on

Hi, I am trying to access a fileserver(NT4 based from a client machine (Windows 2k Pro) and using SAMBA (version 3.0.2a) as a PDC through username raj.It shows me the correct share on the file server but doesn't allow me to change properties of any folder even though I am the admin. The releavnt configuration settings is correct.. I suspect that it gives me guest ACL. When I see my debug 10

Hi List, When we make calls into asterisk with the help of our mobile, landline number, Cisco 79XX series then we didn't able to here any IVR which is playing into asterisk server. But when we dial from SIP softphone then all is working fine and we are able to here the IVR sound files. What is the problem in this case please help me.. -- ----- Thanks and regards Virendra Bhati

Hi everybody We have a strange problem with several asterisk servers (Version 1.4.11) using PRI cards (tied to telco here in Belgium). Indeed we noticed that whenever a local user places an outgoing call through the PRI (and telco) to another IPBX (tied to telco using BRI or PRI), if the remote party places the call on hold, the caller hears the _local_ music on hold instead of the