similar to: How to set outgoing sip callid ?

Hello everyone Is there a possibility to catch an outbound callleg ID for the follovong scenario: some carrier -----> ------(asterisk1) --->-----asterisk2 ? I can get inbound callid for asterisk1 with a ${SIPCALLID} in extensions.conf or to look it up in cdrs field (are the same). But how about outbound? I have all calls just forwarded through asterisk1, not answered and for every call I

Hi, To combine two sources of CDR's I want Asterisk to save the SIP callid for all calls. I know there's a variable that contains the SIP CallID value, but is this the callid value of the incoming INVITE message or the outgoing message? Are they the same? (I've not yet checked a trace, I'm sorry for that). I've tried to read chan_sip, but couldn't find something in the

I am dying here with linphone (not sure if it is crap software or just me being an idiot) but out of the box debian installations of two linphones fail with a "Got SIP response 415 "Unsupported Media Type" back from 192.168.1.3" Can anybody recommend a particular SIP soft phone that broadly satisfies the following criteria? 1. Run on linux. 2. Simple to use and setup. 3. Is

I need to reliably boot a server from a USB device. Since USB device ordering can be unpredictable, I wrote a simple early-userspace "init" program to find the root FS on the correct USB device & partition. I have "labeled" the root ext2 FS on the USB drive, and the program searches for the label by examining the bytes at a specific offset from the beginning of the

Hello All, I've got the following warning messages a couple of days back: /chan_sip.c: Maximum retries exceeded on transmission <SIPcallId> for seqno 1 (Critical Response). /Have got the warnings repeatedly for one Callid. If maximum retries have exceeded why should it give me those warnings again n again for the same callid, with a gap 4 seconds between each warning. The callids

Hello all, I tried the dev list, but got no answer at all. I'm facing some problems with call-id generation in a heavily loaded Asterisk Server. Asterisk is generating same call-id and from tag for different calls (and this is not desirable). Looking at the source code I noticed that rand() is used four times to get a callid. Is that safe enough? Maybe my system lacks of a good random

Hello when I set qualify = yes on trunk I can't do outgoing call. Incoming is always working. [Feb 15 23:01:41] WARNING[12909][C-00000012]: app_dial.c:2525 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) but my linphone is registered all the time. when set qualify = no outgoing call is working (but i have problems when WAN IP is changed after

Hi, I am making calls from the dial plan using the dial application. Due to technical requirements I need to find out the sip call-id used in the dialog initiated by the dial application. I dont see any straight forward way of doing this so I am looking for answers. There is a sip callid session variable but the problems is that dial is a blocking call and the dialog ends when dial returns. I

I am not getting any caller Id with my standard T-1. Is a standard "T" capable of sending callerid? I don't want to spend time troubleshooting my PBX if Asterisk can't send it down that type of trunk. Jordan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, as a followup to my previous posting (re: not getting caller id to display on a powertouch 350), I've found the following... If I hook the PT350 up to the PSTN line, caller id is diplayed properly... If I connect it to our CAS Acess Bank II, the 350 will not display incoming CID... If I hook up a cheap callerid enabled phone to the same port that the PT350 was on (on the AB II), the

Hi I sent this 3 hours ago, seems not go through, so sent again. I have an asterisk php-agi application. It answer's call , then outdial to another number: $agi->exec_dial("SIP", 12345 at test.com , "20", $options); How can I get a SIPCALLID for this out-dialed call? The SIPCALLID seems the incoming call's SIPCALLID. Thanks. Mike

Hey all, I've been fighting with this all morning, and I feel like this should be a relatively simple task, but I just can't get it to work. I currently have a very basic asterisk v11.6 setup with a single extension (a Bria softphone) and a single sip trunk to my carrier. What I'm trying to accomplish is simply adding the asterisk generated SIPCALLID of the leg between asterisk and

Hi all, with an (attended) transfer i see the following happening: 1) A calls B1 2) B2 calls C 3) B2 transfers call to A 4) A talks to C At step 3, the channel A is connected to channel C and B1 and B2 are hung up. In the h extension for channel B2, the channel is renamed to B2<ZOMBIE> and i see that the channel variables of A have been merged into B2<ZOMBIE>. If there were

Hello I have the following scenario: [mynicecontext] exten => 2000,1,Dial(SIP/deviceA&SIP/deviceB&SIP/deviceC) As expected, by dialing 2000, all three devices will ring. And that's fine. However, there are situations where I only want "deviceA" and "deviceB" to ring. I would like to have an extension to dial in order to modify the dialplan. Here is what I

Hello guys, I was some years ago when I sent something to this list. NUT is a great tool but lacks SNMP management. It can access a device using SNMP but I found no way to read NUT values using SNMP. It would be very useful in order to integrate with network management software based on SNMP. As I did not find a solution, I wrote it myself. I wrote a MIB file and a SNMP agent for net-snmp

Hello guys, I was some years ago when I sent something to this list. NUT is a great tool but lacks SNMP management. It can access a device using SNMP but I found no way to read NUT values using SNMP. It would be very useful in order to integrate with network management software based on SNMP. As I did not find a solution, I wrote it myself. I wrote a MIB file and a SNMP agent for net-snmp

Hello. I have two records in dialplan: exten => testA,1,Dial(PJSIP/outgoing/sip:thetestcall at sip.linphone.org) exten => testB,1,Dial(PJSIP/outgoing/sip:thetestcall at iptel.org) Calling testA works fine while testB fails with "CONGESTION". Adding debug for console shows that pjsip_resolver.c does `New queries added, performing parallel resolution again` for linphone after

Hi. I've discovered a bug in the Dial() string processing (for Asterisk 13.14.1 at least). According to the documentation in channels/chan_sip.c the Dial() string syntax is: * SIP/devicename * or SIP/username at domain (SIP uri) * or SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port] * or SIP/devicename/extension * or SIP/devicename/extension/IPorHost * or

It seems a new Linphone 4.2 is to be published next week ! Hopefully, ... Le ven. 5 juin 2020 à 13:34, John Hughes <john at calva.com> a écrit : > On 26/05/2020 15:33, Olivier wrote: > > Hi John, > > 1. Could you get any further, in your quest for working BLF with linphone ? > > The patches to get linphone-3.12 BLF working with Asterisk are here: > >

All the aforementioned techniques need change everytime on the dialplan. I need the office secretary to edit a file (call file) and place it in a particular folder in their windows PCs. this folder is the outgoing folder of LINUX shared through samba in LAN. i need to make it as easy as possible, please. On Tue, Mar 20, 2018 at 5:41 PM, Frank Vanoni <mailinglist at linuxista.com> wrote: