similar to: chan_local

I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed the WIKI page on setting it up but I can't seem to get it to work. Here is my Asterisk version: pbx1*CLI> core show version Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on 2008-01-10 12:08:48 UTC Here is a log of when the FollowMe is being called: NOTE: I've tried to use the AstDB as

Hi guys, i just came to know that CDR(dst) field is set to current extension instead of the dialed no. i need to set it to DNID because our every user has 5 dids and i want to show the caller at the end of the month which numbers he dialed for every call, along with other cdr info. Our rating depends on the dialed number also. here is my extensions.conf exten=> 1212,1,Dial(SIP/rizwan)

hi all, i just finished developing some incoming call features in a macro. that macro gets executed everytime an incoming call is received and a new mysql connection is made using the MYSQL cmd in dialplan. i want to use a single mysql connection for every incoming call. my idea of doing it is like this, i want to get a mysql connection in a global variable, just to share the connection with

Hi all, I need a fully developed web based remote provisioning system. I cant find anything reliable on the internet. Have already checked ataconfig.com and voxilla-ays.com. have tried to contact them but got no response. So if anybody knows a good provisioning system then plz tell me about it. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part

Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

im having a strange problem related to call-limit for peers. well im not sure if its related to call-limmit or not. Bottom line is: I call a user A, from user B. user B hears silence, untill it goes to voicemail. when user B hangsup. user B's call limit is reset to 0 but user A's call limit is not reset.strange thing is user A's status on cli is shown as NOANSWER, while user B did not

Hi all, Is there anyway i can prevent multiple sip registrations from different IPs using single username in asterisk. Does asterisk provide any aid in this respect? As far as my knowledge is concerned i dont think there is any support for this in asterisk, so i think i'll have to makeup a script which sniffs sip packets coming for asterisk and detect for multiple register requests coming from

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ CLI Output : ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vicidialnow*CLI> == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from

Hi all, i need an XML file format which is used in remote provisioning of different spa devices. Please somebody tell me the format or tell me where can i find it on the internet. I also need a list of parameters which are configured using auto-provisioning. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was

Hi all, i need to study the SIP protocol. can anybody tell me about any ebook which could halp me understand the sip protocol, architecture, and how to read and understand the sip signalling when i use "sip debug" in asterisk? -- Rizwan Hisham Software Engineer AXVOICE Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, We are planning to shift our sip users from one platform to another. (basically from one asterisk server to another). the problem we are facing is that both asterisk servers are using different ports to listen for sip. and both have live customers on them. provisioning their ata's is not a good option for us coz of our settup. we cant just ask the customers to change their ports for

Hello, I have a question . dose asterisk supports "302 Redirecting..." ? I have SIP Server "Not Asterisk" and my Asterisk is registering as a client for this device . when i try to call another client registered to the same SIP server i got Busy Tone and here is the asterisk CLI output ----------------- -- Got SIP response 302 "Redirecting..." back

Hi all, Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? -- Best Regards

Hi all, Im trying dial a user according to the device s/he uses. i mean if the user is using asterisk as a peer, then i have to pass the extension in the dial application like this: Dial(SIP/${EXTEN}@user) ;so that s/he can perform routing according to the DNID. and if the user is using sipura, linksys or grandstream i dial the user like this, Dial(SIP/user) so is there a way to know what kind

Hi all, I am using an asterisk as a client to connect to another asterisk server by registering with the register string. Registration is done without any hassel, but after sometime my asterisk loses the registration with the server and the server starts displaying the following msgs repeatedly: [Aug 9 06:37:59] NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce

Hi all, There is a parameter called "nonce" included in every register request that a UA sends to asterisk. I have read sip debug a lot and only found out that the "nonce" parameter value which is used in register request was generated by asterisk server in a previous sip response. As you can see in the sip debug (labled in red). <--- Transmitting (NAT) to

Hi folks, this does look like a bug to me: Asterisk replaces the @63.214.186.6 by @context which obviously leads to a failure. Any comments, do I have a configuration issue on my side that I missed? Cheers, Philipp -- Executing Dial("SIP/philipp-bd5f", "SIP/992365264680@nikotel- out|90") in new stack -- Called 99xxxxxxxxxx@nikotel-out -- Got SIP response 302

Hello fellow asterisk people! I have Asterisk listening on port 5061 and SER on port 5060. Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP. My problems are with SIP. I can make incoming calls from SIP to asterisk and to any of the other networks, but when I try to make an outgoing call from Asterisk to SER I see the following in Asterisk: -- Executing

I am migrating my app to Asterisk12 and pjsip, but I cannot find chan_local, what happened?

I'm having some difficulty with chan_local dial requests. It seems that when a chan_local call is picked up, that the native bridge "pops" the environment back to the settings of the original call. This is unexpected and leads to very frustrating results. My example below is a very distilled sample of a much more complex dialplan problem I'm having with chan_local, but it