similar to: SIPCALLID, but which 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

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

Calling linphone from asterisk 13.9.1.: Dial(SIP/<user>@sip.linphone.org) And it works. But on the linphone side the caller is: <extno>@ipaddress or 2502 at 45.123.987.4 Is there any way to make it more descriptive, at least for the sip user name ? I tried setting SIPCALLID, which had no effect. Set(SIPCALLID=Office) Thanks, sean

> I have seen the following effect in Asterisk, though: where > it converts > an inband DTMF (eg coming off a Zap channel) into an > indication, it mutes > the audio where that tone is. But sometimes it leaves a > teeny bit of the > tone behind. > > If you take such a call over say IAX to somewhere and then > back out a Zap > channel, you end up with the

Hello Anyone have a working copy of Fail2ban asterisk filter asterisk.conf for Asterisk 16 running PJSIP. I have tried 10 different filters but none of them show any matches when testing with fail2ban-regex I see date template hits but no matches.... My log [2019-06-06 15:37:20] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" <sip:2405 at

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, Does anybody know how to get the SIP call ID of a "Dial" command? Thanks in advance. Ray -- Want an e-mail address like mine? Get a free e-mail account today at www.mail.com! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071008/7f27548e/attachment.htm

Has anyone else had problems with users being able to press key tones during a voice prompt? I have a few users complaining that some systems will not recognize key presses during them. using current CVS-HEAD, linksys PAP2 UA's, rfc2833 dtmf mode. Thanks Sherwood McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Sirs, According to RFC 2705 (MGCP), these are the parameters that are used in the transactions: ReturnCode, Connection-parameters <-- DeleteConnection(CallId, EndpointId, ConnectionId, [Encapsulated NotificationRequest,] [Encapsulated

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

Hello, I tried to configure a very simple case of Asterisk using SIP userA --- Asterisk server ---- userB sip.conf [userA] type=friend username=userA host=dynamic nat=no context=test [userB] type=friend username=userB host=dynamic nat=no context=test In extensions.conf [test] exten => 1000,1,Dial(SIP/userA) exten => 2000,1,Dial(SIP/userB) I make a call from userA to userB, it works,

> Next step is packet capture to see if there is a clue as to the cause of the > failure in the SIP signalling. Right, I see the following when running SIP Debug. Looks to me like the phones are expecting the server to do the conference mixing, which I guess it would do in CallManager? <--- SIP read from TCP:xxx.xxx.xxx.xxx:50604 ---> REFER sip:xxx.xxx.xxx.xxx SIP/2.0 Via:

Hi, In a VoIP only environment, Asterisk has to use ztdummy to have any chance of playing back understandable audio files (without drops, hickups etc). I have been using ztdummy to some degree of success, but I also have a "Wildcard TDM400P REV E/F Board 1" in the Asterisk machine I'm using. I'm not using this card for anything at all, but I'm wondering how to set it

No Jeremy, excuse me, the error was in my email. The correct command is /bin/echo "Channel: Local/$1@chiamamezzi-dialout";\ /bin/echo "Variable: callid=123456|number=$1|url=pippo|menuid=FOP|redirectnum=0554202880";\ /bin/echo "Context: chiamamezzi-Wave";\ /bin/echo "Exten: s";\ /bin/echo "Priority: 1";\ /bin/echo "Callerid: Asterisk Automatic

Hi all, I am using Digium PCI board to receive PSTN call through regular phone line. It is no problem for me to receive calls, but I am not able to obtain the Caller ID if the calls are from the phone line. exten => s,1,Answer() exten => s, n, Verbose(1|incoming number is ${CHANNEL} calling to ${EXTEN} routing to ${phonenum} ) exten => s,n, Verbose(1|callid is ${CALLID(num)}) exten

Hello! Upgraded 13.10 to 13.11.1 today and now I see messages in log: [Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"3563" <sip:3563 at 192.168.32.254>' failed for '192.168.32.116:5060' (callid: 0_1409534529 at 192.168.32.116) - No matching endpoint found or [Sep 9 12:56:14] NOTICE[10163]

09.09.2016 13:45, Joshua Colp ?????: > Dmitry Melekhov wrote: >> Hello! >> >> >> Upgraded 13.10 to 13.11.1 today and now I see messages in log: >> >> >> [Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request >> 'REGISTER' from '"3563" <sip:3563 at 192.168.32.254>' failed for >>

Hi, We've got endpoints and gateways who have T.38 fax support. We now use SER and Asterisk to do our routing and other functionality, but fax doesn't seem to work. Asterisk complains like this: Sep 9 09:25:45 WARNING[467828746]: RTP Read too short Sep 9 09:25:45 WARNING[467828746]: Asked to transmit frame type 256, while native formats is 32 (read/write = 256/256) With lots of the

asterisk-users-bounces@lists.digium.com wrote: > Isn't it possible to use T.38 for interconnecting hardware gates > supporting T.38 with asterisk using SIP REINVITE? > I'm not shure but but think its's might be possible because after > reinvite traffic goes directly from one gate to anotger, not over > Asterisk We've seen a problem here with asterisk. Wehn

Hello there! I'm working on some modifications on Asterisk to adapt it to our needs considering some particular demandings of the infraestructure we want to provide. Two of these modifications are: 1- A proprietary configuration driver that will communicate with a server that will be the source of information for the entire infraestructure; and, 2- A call control application that will be