[asterisk-users] Trunking betweeb two Asterisk System
Hi guys, I am trying to make a trunk between two asterisk system SIP Trunk on Asterisk 1.6 but I cannt make it work, can any body help me plz? Thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Executing Script after MixMonitor is called
Hello Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav" "/var/spool/asterisk/monitor/${CALLFILENAME}.mp3" && rm -f "/var/spool/asterisk/monitor/${CALLFILENAME}.wav") exten=6500,n,MixMonitor(${CALLFILENAME}.wav,b) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change the caller's phone number
or this Set(${CALLERID(all)}="722979797" <722979797>) From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Thursday, January 19, 2012 8:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Change the caller's phone number try the following Set(${CALLERID}="722979797" <722979797>) From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal [e...@mcr-m.com] Sent: Thursday, January 19, 2012 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Change the caller's phone number Hi, I have a system that receives calls from clients and directs them to an external phone, before I pass on the client I change the client's phone number to a number that I choose, so that The call recipient knew the call came from our system. But I have a problem with that, not all phone number change some of the anonymous calls stay anonymous and the recipient See in the caller ID display unlisted number. I use this commend: Set(CALLERID(all)="722979797" <722979797>) Is anyone having a similar problem or know what the problem? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change the caller's phone number
try the following Set(${CALLERID}="722979797" <722979797>) From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal [e...@mcr-m.com] Sent: Thursday, January 19, 2012 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Change the caller's phone number Hi, I have a system that receives calls from clients and directs them to an external phone, before I pass on the client I change the client's phone number to a number that I choose, so that The call recipient knew the call came from our system. But I have a problem with that, not all phone number change some of the anonymous calls stay anonymous and the recipient See in the caller ID display unlisted number. I use this commend: Set(CALLERID(all)="722979797" <722979797>) Is anyone having a similar problem or know what the problem? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Call type in dial plan
I already tried what u posted didnt work but thanx for the reply :) From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind [govoi...@gmail.com] Sent: Wednesday, January 04, 2012 11:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call type in dial plan Hi, Sorry for late reply. Hope you've already found out something about it. What version of asterisk you are using, that function for choosing inbound/outbound call leg codecs is for newer versions of asterisk. See these pages: http://www.voip-info.org/wiki/view/Asterisk+variables https://issues.asterisk.org/view.php?id=13243 Regards, Sammy On Tue, Jan 3, 2012 at 2:31 PM, Faraj Khasib mailto:fkha...@iconnecths.com>> wrote: thats excatly what I want, can u plz give me the command, I want to choose only ulow From: asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com> [asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>] On Behalf Of Sammy Govind [govoi...@gmail.com<mailto:govoi...@gmail.com>] Sent: Tuesday, January 03, 2012 3:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call type in dial plan Hi, For such call you just need to select the outbound codec before the dial() app. choose the audio-only codecs and thus no video codec strings will be exchanged in that call. -- Regards, Sammy On Tue, Jan 3, 2012 at 1:54 PM, Faraj Khasib mailto:fkha...@iconnecths.com><mailto:fkha...@iconnecths.com<mailto:fkha...@iconnecths.com>>> wrote: this is what my SIP Invite message when I make Video call INVITE sip:6500@192.168.21.102<mailto:sip%3A6500@192.168.21.102><mailto:sip%3A6500@192.168.21.102<mailto:sip%253A6500@192.168.21.102>> SIP/2.0 Via: SIP/2.0/UDP 192.168.21.193:52933;branch=z9hG4bK1943005978;rport From: mailto:sip%3A6097@192.168.21.102><mailto:sip%3A6097@192.168.21.102<mailto:sip%253A6097@192.168.21.102>>>;tag=1857098215 To: mailto:sip%3A6500@192.168.21.102><mailto:sip%3A6500@192.168.21.102<mailto:sip%253A6500@192.168.21.102>>> Contact: ;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" Call-ID: b9453704-d76a-b8ce-3247-c999abff7395 CSeq: 324677463 INVITE Content-Type: application/sdp Content-Length: 588 Max-Forwards: 70 Route: Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER Privacy: none P-Access-Network-Info: ADSL;utran-cell-id-3gpp= User-Agent: Medcor Supported: 100rel v=0 o=doubango 1983 678901 IN IP4 192.168.21.193 s=- c=IN IP4 192.168.21.193 t=0 0 m=audio 36372 RTP/AVP 8 0 9 101 a=ptime:20 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:9 G722/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 m=video 59296 RTP/AVP 125 106 121 103 a=rtpmap:125 VP8/9 a=fmtp:125 CIF=2;QCIF=2;SQCIF=2 a=rtpmap:106 H264/9 a=fmtp:106 profile-level-id=42e01e; packetization-mode=1; max-br=452; max-mbps=11880 a=rtpmap:121 MP4V-ES/9 a=fmtp:121 profile-level-id=3 a=rtpmap:103 H263-1998/9 a=fmtp:103 CIF=2;QCIF=2;SQCIF=2 when I make Audio call requests I dont have the video part but at receiver since two clients can make video call they have Asterisks adds the Video Part in request sent to receiver,I dont want that part added , how I can delete it ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Call Codec in extension.conf
yup and video support is yes From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 12:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf Both sides? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf allow=all From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 12:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf What about the allow/disallow lines in sip.conf? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Call Codec in extension.conf
allow=all From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 12:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf What about the allow/disallow lines in sip.conf? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Call Codec in extension.conf
there is nothing in sip.conf about what u asked but 6500 is a queue with following info [6500] fullname = testing strategy = rrmemory timeout = 15 wrapuptime = 15 autofill = no autopause = no joinempty = yes leavewhenempty = no reportholdtime = no maxlen = 0 musicclass = test member = SIP/6251 member = SIP/6252 member = SIP/6253 member = SIP/6254 now the user 6251 is a user with following info and caller 6000 [6000] username = 6000 transfer = yes mailbox = 6000 call-limit = 100 type = peer fullname = 6000 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 6000 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = yes nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = ulaw,gsm,h263,h263p,h264 autoprov = no label = macaddress = linenumber = 1 LINEKEYS = 1 callcounter = yes [6251] username = 6251 transfer = yes mailbox = 6251 call-limit = 100 type = peer fullname = 6251 registersip = no host = dynamic callgroup = 1 type = peer context = DLPN_DialPlan1 cid_number = 6251 hasvoicemail = no vmsecret = email = threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = yes hassip = yes hasiax = yes nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = ulaw,gsm,h263,h263p,h264 autoprov = no label = macaddress = linenumber = 1 LINEKEYS = 1 From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 12:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf Please post the sip.conf entries for 6000 and 6500. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf I am the other end most codecs are available now my problem is when I make audio call using one side its converted to video call request (since my other end has also all codecs) my app clients can do Audio and Video call, now the Video call is ok but the Audio part get converted to video request ...so I am trying to limit the codec to only audio codec... From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:54 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf You are fighting a losing battle - you can't control the other end Ignoring ${SIP_CODEC} variable because it is not shared by both ends. You can probably do a SIP SET DEBUG ON and see what codecs are available on the other end. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf -- Executing [6500@DLPN_DialPlan1:1] Set("SIP/6000-", "SIP_CODEC=gsm ") in new stack -- Executing [6500@DLPN_DialPlan1:2] Set("SIP/6000-", "SIP_CODEC_INB OUND=gsm") in new stack -- Executing [6500@DLPN_DialPlan1:3] Set("SIP/6000-", "SIP_CODEC_OUT BOUND=gsm") in new stack -- Executing [6500@DLPN_DialPlan1:4] Answer("SIP/6000-", "") in new stack [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. -- Executing [6500@DLPN_DialPlan1:5] Playback("SIP/6000-", "welcome" ) in new stack [Jan 4 17:50:16] WARNING[4131]: file.c:650 ast_openstream_full: File welcome do es not exist in any format [Jan 4 17:50:16] WARNING[4131]: file.c:953 ast_streamfile: Unable to open welco me (format 0x4 (ulaw)): No such file or directory [Jan 4 17:50:16] WARNING[4131]: app_playback.c:471 playback_exec
Re: [asterisk-users] Set Call Codec in extension.conf
Any suggestion will be great From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Wednesday, January 04, 2012 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf I am the other end most codecs are available now my problem is when I make audio call using one side its converted to video call request (since my other end has also all codecs) my app clients can do Audio and Video call, now the Video call is ok but the Audio part get converted to video request ...so I am trying to limit the codec to only audio codec... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Call Codec in extension.conf
I am the other end most codecs are available now my problem is when I make audio call using one side its converted to video call request (since my other end has also all codecs) my app clients can do Audio and Video call, now the Video call is ok but the Audio part get converted to video request ...so I am trying to limit the codec to only audio codec... From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:54 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf You are fighting a losing battle - you can't control the other end Ignoring ${SIP_CODEC} variable because it is not shared by both ends. You can probably do a SIP SET DEBUG ON and see what codecs are available on the other end. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf -- Executing [6500@DLPN_DialPlan1:1] Set("SIP/6000-", "SIP_CODEC=gsm ") in new stack -- Executing [6500@DLPN_DialPlan1:2] Set("SIP/6000-", "SIP_CODEC_INB OUND=gsm") in new stack -- Executing [6500@DLPN_DialPlan1:3] Set("SIP/6000-", "SIP_CODEC_OUT BOUND=gsm") in new stack -- Executing [6500@DLPN_DialPlan1:4] Answer("SIP/6000-", "") in new stack [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. -- Executing [6500@DLPN_DialPlan1:5] Playback("SIP/6000-", "welcome" ) in new stack [Jan 4 17:50:16] WARNING[4131]: file.c:650 ast_openstream_full: File welcome do es not exist in any format [Jan 4 17:50:16] WARNING[4131]: file.c:953 ast_streamfile: Unable to open welco me (format 0x4 (ulaw)): No such file or directory [Jan 4 17:50:16] WARNING[4131]: app_playback.c:471 playback_exec: ast_streamfil e failed on SIP/6000- for welcome -- Executing [6500@DLPN_DialPlan1:6] SIPAddHeader("SIP/6000-", "emai l:fkha...@iconnecths.com") in new stack -- Executing [6500@DLPN_DialPlan1:7] MixMonitor("SIP/6000-", "2012-0 1-05//2012-01-05_05:50:16_thursday_fkha...@iconnecths.com.wav,b") in new stack -- Executing [6500@DLPN_DialPlan1:8] Queue("SIP/6000-", "6500") in n ew stack -- Started music on hold, class 'default', on SIP/6000- == Begin MixMonitor Recording SIP/6000- From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf CLI output from call? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf didnt work also :( From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is cumbersome; 1-n-n-n-n-n is more practical). Like this exten=6500,1,Answer exten=6500,n,Playback(welcome) exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,n,Queue(${EXTEN}) or exten=6500,1,set(SIP_CODEC=gsm) -- this is not changed exten=6500,n,Answer exten=6500,n,Playba
Re: [asterisk-users] Set Call Codec in extension.conf
-- Executing [6500@DLPN_DialPlan1:1] Set("SIP/6000-", "SIP_CODEC=gsm ") in new stack -- Executing [6500@DLPN_DialPlan1:2] Set("SIP/6000-", "SIP_CODEC_INB OUND=gsm") in new stack -- Executing [6500@DLPN_DialPlan1:3] Set("SIP/6000-", "SIP_CODEC_OUT BOUND=gsm") in new stack -- Executing [6500@DLPN_DialPlan1:4] Answer("SIP/6000-", "") in new stack [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. -- Executing [6500@DLPN_DialPlan1:5] Playback("SIP/6000-", "welcome" ) in new stack [Jan 4 17:50:16] WARNING[4131]: file.c:650 ast_openstream_full: File welcome do es not exist in any format [Jan 4 17:50:16] WARNING[4131]: file.c:953 ast_streamfile: Unable to open welco me (format 0x4 (ulaw)): No such file or directory [Jan 4 17:50:16] WARNING[4131]: app_playback.c:471 playback_exec: ast_streamfil e failed on SIP/6000- for welcome -- Executing [6500@DLPN_DialPlan1:6] SIPAddHeader("SIP/6000-", "emai l:fkha...@iconnecths.com") in new stack -- Executing [6500@DLPN_DialPlan1:7] MixMonitor("SIP/6000-", "2012-0 1-05//2012-01-05_05:50:16_thursday_fkha...@iconnecths.com.wav,b") in new stack -- Executing [6500@DLPN_DialPlan1:8] Queue("SIP/6000-", "6500") in n ew stack -- Started music on hold, class 'default', on SIP/6000- == Begin MixMonitor Recording SIP/6000- From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf CLI output from call? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf didnt work also :( From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is cumbersome; 1-n-n-n-n-n is more practical). Like this exten=6500,1,Answer exten=6500,n,Playback(welcome) exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,n,Queue(${EXTEN}) or exten=6500,1,set(SIP_CODEC=gsm) -- this is not changed exten=6500,n,Answer exten=6500,n,Playback(welcome) exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,n,Queue(${EXTEN}) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf how can u give me a command?!.. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:29 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf My guess is that you should set the codec either before SIPADDHEADER or before ANSWER. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [
Re: [asterisk-users] Set Call Codec in extension.conf
didnt work also :( From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is cumbersome; 1-n-n-n-n-n is more practical). Like this exten=6500,1,Answer exten=6500,n,Playback(welcome) exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,n,Queue(${EXTEN}) or exten=6500,1,set(SIP_CODEC=gsm) -- this is not changed exten=6500,n,Answer exten=6500,n,Playback(welcome) exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,n,Queue(${EXTEN}) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf how can u give me a command?!.. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:29 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf My guess is that you should set the codec either before SIPADDHEADER or before ANSWER. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf I tried also in asterisk 1.8 setting outbound variable but didnt work also https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables check the above ... I changed it and tried but still I get a video call From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 does not support setting the outbound codec.1.8 uses different variables to set the outbound codec. See UGRADE.txt in the Asterisk source for the 1.8 information,. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 and 1.8 ... I tried changing stuff on both when I make audio call from my client which supports both audio and video its sent to the other client as video call .I tried settings the SIP_CODED_INBOUND and outbound also ... but no luck From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf Providing which version of Asterisk you are using might be helpful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf anyhelp guys? I tried a lot of stuff but it doesnt work the Codec for audio call only cannt be set...how I can set the call type video/audio at dail plan? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Set Call Codec in extension.conf Hi All, I am trying to set call codec at extension.conf but it doesnt work ... its like my command doesnt change anything exten=6500,1,Answer exten=6500,2,Playback(welcome) exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)}) ext
Re: [asterisk-users] Set Call Codec in extension.conf
how can u give me a command?!.. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:29 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf My guess is that you should set the codec either before SIPADDHEADER or before ANSWER. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf I tried also in asterisk 1.8 setting outbound variable but didnt work also https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables check the above ... I changed it and tried but still I get a video call From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 does not support setting the outbound codec.1.8 uses different variables to set the outbound codec. See UGRADE.txt in the Asterisk source for the 1.8 information,. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 and 1.8 ... I tried changing stuff on both when I make audio call from my client which supports both audio and video its sent to the other client as video call .I tried settings the SIP_CODED_INBOUND and outbound also ... but no luck From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf Providing which version of Asterisk you are using might be helpful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf anyhelp guys? I tried a lot of stuff but it doesnt work the Codec for audio call only cannt be set...how I can set the call type video/audio at dail plan? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Set Call Codec in extension.conf Hi All, I am trying to set call codec at extension.conf but it doesnt work ... its like my command doesnt change anything exten=6500,1,Answer exten=6500,2,Playback(welcome) exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed exten=6500,6,Queue(${EXTEN}) can any body help me with that? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/aste
Re: [asterisk-users] Set Call Codec in extension.conf
I tried also in asterisk 1.8 setting outbound variable but didnt work also https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables check the above ... I changed it and tried but still I get a video call From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 does not support setting the outbound codec.1.8 uses different variables to set the outbound codec. See UGRADE.txt in the Asterisk source for the 1.8 information,. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 and 1.8 ... I tried changing stuff on both when I make audio call from my client which supports both audio and video its sent to the other client as video call .I tried settings the SIP_CODED_INBOUND and outbound also ... but no luck From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf Providing which version of Asterisk you are using might be helpful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf anyhelp guys? I tried a lot of stuff but it doesnt work the Codec for audio call only cannt be set...how I can set the call type video/audio at dail plan? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Set Call Codec in extension.conf Hi All, I am trying to set call codec at extension.conf but it doesnt work ... its like my command doesnt change anything exten=6500,1,Answer exten=6500,2,Playback(welcome) exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed exten=6500,6,Queue(${EXTEN}) can any body help me with that? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Call Codec in extension.conf
1.6 and 1.8 ... I tried changing stuff on both when I make audio call from my client which supports both audio and video its sent to the other client as video call .I tried settings the SIP_CODED_INBOUND and outbound also ... but no luck From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@nyigc.com] Sent: Wednesday, January 04, 2012 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf Providing which version of Asterisk you are using might be helpful. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf anyhelp guys? I tried a lot of stuff but it doesnt work the Codec for audio call only cannt be set...how I can set the call type video/audio at dail plan? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Set Call Codec in extension.conf Hi All, I am trying to set call codec at extension.conf but it doesnt work ... its like my command doesnt change anything exten=6500,1,Answer exten=6500,2,Playback(welcome) exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed exten=6500,6,Queue(${EXTEN}) can any body help me with that? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Call Codec in extension.conf
anyhelp guys? I tried a lot of stuff but it doesnt work the Codec for audio call only cannt be set...how I can set the call type video/audio at dail plan? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Set Call Codec in extension.conf Hi All, I am trying to set call codec at extension.conf but it doesnt work ... its like my command doesnt change anything exten=6500,1,Answer exten=6500,2,Playback(welcome) exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed exten=6500,6,Queue(${EXTEN}) can any body help me with that? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Set Call Codec in extension.conf
Hi All, I am trying to set call codec at extension.conf but it doesnt work ... its like my command doesnt change anything exten=6500,1,Answer exten=6500,2,Playback(welcome) exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed exten=6500,6,Queue(${EXTEN}) can any body help me with that? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to make SIP guest call
thanx alot ... :) that helped From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Roland [aster...@rolandow.com] Sent: Tuesday, January 03, 2012 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to make SIP guest call I managed to do that once by using another SIP account, for example at Voipbuster. It's free. Once you are connected, you can still use ext@IP of your server. I guess you could use any other free SIP account. On Tue, Jan 3, 2012 at 4:01 PM, Faraj Khasib mailto:fkha...@iconnecths.com>> wrote: thank you for your reply, but x-lite cannt dail without an active account dail is disabled without any account My problem is that I am trying to have multiclients call my SIP queue, now each client is not authorized so I tried to make them call using the same extension but I got call overlap between all clients, now what I want is a way that I can make all my client call the SIP queue using SIP protocol, I am thinking using guest Call it would solve my problem, can u plz help me, if you have any other suggestion plz do Thanx From: asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com> [asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>] On Behalf Of Patrick Lists [asterisk-l...@puzzled.xs4all.nl<mailto:asterisk-l...@puzzled.xs4all.nl>] Sent: Tuesday, January 03, 2012 8:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to make SIP guest call On 03-01-12 14:13, Faraj Khasib wrote: > anyone? > what should x-lite account be for guest user ?I tried guest but didnt work A guest does not need an account on your asterisk server so you do not need to configure an account on xlite. Instead on xlite you just dial @ Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to make SIP guest call
thank you for your reply, but x-lite cannt dail without an active account dail is disabled without any account My problem is that I am trying to have multiclients call my SIP queue, now each client is not authorized so I tried to make them call using the same extension but I got call overlap between all clients, now what I want is a way that I can make all my client call the SIP queue using SIP protocol, I am thinking using guest Call it would solve my problem, can u plz help me, if you have any other suggestion plz do Thanx From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Lists [asterisk-l...@puzzled.xs4all.nl] Sent: Tuesday, January 03, 2012 8:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to make SIP guest call On 03-01-12 14:13, Faraj Khasib wrote: > anyone? > what should x-lite account be for guest user ?I tried guest but didnt work A guest does not need an account on your asterisk server so you do not need to configure an account on xlite. Instead on xlite you just dial @ Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Registering multi-clients
hey all, My problem is that I am trying to have multiclients call my SIP queue, now each client is not authorized so I tried to make them call using the same extension but I got call overlap between all clients, now what I want is a way that I can make all my client call the SIP queue using SIP protocol, I am thinking using guest Call it would solve my problem, can u plz help me, if you have any other suggestion plz do Thanx -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to make SIP guest call
anyone? what should x-lite account be for guest user ?I tried guest but didnt work From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Tuesday, January 03, 2012 5:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to make SIP guest call for example if I am using x-lite as client, how to I connect as guest from client ...I am allowing guests at asterisk server From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Tuesday, January 03, 2012 5:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to make SIP guest call thank you very much for this explanation, but my question does my client have to be registered first, right? what do i Use to register ... there should be information to register with using guest, I got your idea about the security, and I can work with that ... but at cleints I need to have information to log with?right? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of David Klaverstyn [da...@klaverstyn.com.au] Sent: Tuesday, January 03, 2012 5:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to make SIP guest call Hi, The point of SIP guest calls is that there is no username and password required to make calls. If you have enabled guest calls then whatever extensions you have allowed in the allocate default sip context anyone will be able to dial. If you have in your sip.conf file context=from-vsp; Default context for incoming calls allowguest=yes ; Allow or reject guest calls (default is yes) and in your extensions.conf file exten => 202,1,GotoIf($[${LEN(${CALLERID(name)})}=0]?2:3) exten => 202,n,Set(CALLERID(NAME)=Guest SIP User) exten => 202,n,Dial(SIP/202,30,r) exten => 202,n,VoiceMail(202@default,us) exten => 202,n,HangUp ... then anyone will be able to call 202. The key is to make sure people cannot make trunk calls from the guest context. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Tuesday, 3 January 2012 9:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to make SIP guest call Actaully I didnt find a good example how to configure the guest call in asterisk other than allowGuest in SIP.conf, anybody have a good example for that? thanx From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Tuesday, January 03, 2012 5:08 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to make SIP guest call Hi all, If I am enabling the SIP Guest calls, How can I make the call? what my SIP clients information to make the call? I mean what there username and password for guest call? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.
Re: [asterisk-users] How to make SIP guest call
for example if I am using x-lite as client, how to I connect as guest from client ...I am allowing guests at asterisk server From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Tuesday, January 03, 2012 5:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to make SIP guest call thank you very much for this explanation, but my question does my client have to be registered first, right? what do i Use to register ... there should be information to register with using guest, I got your idea about the security, and I can work with that ... but at cleints I need to have information to log with?right? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of David Klaverstyn [da...@klaverstyn.com.au] Sent: Tuesday, January 03, 2012 5:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to make SIP guest call Hi, The point of SIP guest calls is that there is no username and password required to make calls. If you have enabled guest calls then whatever extensions you have allowed in the allocate default sip context anyone will be able to dial. If you have in your sip.conf file context=from-vsp; Default context for incoming calls allowguest=yes ; Allow or reject guest calls (default is yes) and in your extensions.conf file exten => 202,1,GotoIf($[${LEN(${CALLERID(name)})}=0]?2:3) exten => 202,n,Set(CALLERID(NAME)=Guest SIP User) exten => 202,n,Dial(SIP/202,30,r) exten => 202,n,VoiceMail(202@default,us) exten => 202,n,HangUp ... then anyone will be able to call 202. The key is to make sure people cannot make trunk calls from the guest context. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Tuesday, 3 January 2012 9:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to make SIP guest call Actaully I didnt find a good example how to configure the guest call in asterisk other than allowGuest in SIP.conf, anybody have a good example for that? thanx From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Tuesday, January 03, 2012 5:08 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to make SIP guest call Hi all, If I am enabling the SIP Guest calls, How can I make the call? what my SIP clients information to make the call? I mean what there username and password for guest call? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to make SIP guest call
thank you very much for this explanation, but my question does my client have to be registered first, right? what do i Use to register ... there should be information to register with using guest, I got your idea about the security, and I can work with that ... but at cleints I need to have information to log with?right? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of David Klaverstyn [da...@klaverstyn.com.au] Sent: Tuesday, January 03, 2012 5:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to make SIP guest call Hi, The point of SIP guest calls is that there is no username and password required to make calls. If you have enabled guest calls then whatever extensions you have allowed in the allocate default sip context anyone will be able to dial. If you have in your sip.conf file context=from-vsp; Default context for incoming calls allowguest=yes ; Allow or reject guest calls (default is yes) and in your extensions.conf file exten => 202,1,GotoIf($[${LEN(${CALLERID(name)})}=0]?2:3) exten => 202,n,Set(CALLERID(NAME)=Guest SIP User) exten => 202,n,Dial(SIP/202,30,r) exten => 202,n,VoiceMail(202@default,us) exten => 202,n,HangUp ... then anyone will be able to call 202. The key is to make sure people cannot make trunk calls from the guest context. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Tuesday, 3 January 2012 9:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to make SIP guest call Actaully I didnt find a good example how to configure the guest call in asterisk other than allowGuest in SIP.conf, anybody have a good example for that? thanx From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Tuesday, January 03, 2012 5:08 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to make SIP guest call Hi all, If I am enabling the SIP Guest calls, How can I make the call? what my SIP clients information to make the call? I mean what there username and password for guest call? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to make SIP guest call
Actaully I didnt find a good example how to configure the guest call in asterisk other than allowGuest in SIP.conf, anybody have a good example for that? thanx From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Tuesday, January 03, 2012 5:08 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to make SIP guest call Hi all, If I am enabling the SIP Guest calls, How can I make the call? what my SIP clients information to make the call? I mean what there username and password for guest call? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to make SIP guest call
Hi all, If I am enabling the SIP Guest calls, How can I make the call? what my SIP clients information to make the call? I mean what there username and password for guest call? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Call type in dial plan
thats excatly what I want, can u plz give me the command, I want to choose only ulow From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind [govoi...@gmail.com] Sent: Tuesday, January 03, 2012 3:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call type in dial plan Hi, For such call you just need to select the outbound codec before the dial() app. choose the audio-only codecs and thus no video codec strings will be exchanged in that call. -- Regards, Sammy On Tue, Jan 3, 2012 at 1:54 PM, Faraj Khasib mailto:fkha...@iconnecths.com>> wrote: this is what my SIP Invite message when I make Video call INVITE sip:6500@192.168.21.102<mailto:sip%3A6500@192.168.21.102> SIP/2.0 Via: SIP/2.0/UDP 192.168.21.193:52933;branch=z9hG4bK1943005978;rport From: mailto:sip%3A6097@192.168.21.102>>;tag=1857098215 To: mailto:sip%3A6500@192.168.21.102>> Contact: ;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" Call-ID: b9453704-d76a-b8ce-3247-c999abff7395 CSeq: 324677463 INVITE Content-Type: application/sdp Content-Length: 588 Max-Forwards: 70 Route: Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER Privacy: none P-Access-Network-Info: ADSL;utran-cell-id-3gpp= User-Agent: Medcor Supported: 100rel v=0 o=doubango 1983 678901 IN IP4 192.168.21.193 s=- c=IN IP4 192.168.21.193 t=0 0 m=audio 36372 RTP/AVP 8 0 9 101 a=ptime:20 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:9 G722/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 m=video 59296 RTP/AVP 125 106 121 103 a=rtpmap:125 VP8/9 a=fmtp:125 CIF=2;QCIF=2;SQCIF=2 a=rtpmap:106 H264/9 a=fmtp:106 profile-level-id=42e01e; packetization-mode=1; max-br=452; max-mbps=11880 a=rtpmap:121 MP4V-ES/9 a=fmtp:121 profile-level-id=3 a=rtpmap:103 H263-1998/9 a=fmtp:103 CIF=2;QCIF=2;SQCIF=2 when I make Audio call requests I dont have the video part but at receiver since two clients can make video call they have Asterisks adds the Video Part in request sent to receiver,I dont want that part added , how I can delete it ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Call type in dial plan
this is what my SIP Invite message when I make Video call INVITE sip:6500@192.168.21.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.21.193:52933;branch=z9hG4bK1943005978;rport From: ;tag=1857098215 To: Contact: ;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" Call-ID: b9453704-d76a-b8ce-3247-c999abff7395 CSeq: 324677463 INVITE Content-Type: application/sdp Content-Length: 588 Max-Forwards: 70 Route: Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER Privacy: none P-Access-Network-Info: ADSL;utran-cell-id-3gpp= User-Agent: Medcor Supported: 100rel v=0 o=doubango 1983 678901 IN IP4 192.168.21.193 s=- c=IN IP4 192.168.21.193 t=0 0 m=audio 36372 RTP/AVP 8 0 9 101 a=ptime:20 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:9 G722/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 m=video 59296 RTP/AVP 125 106 121 103 a=rtpmap:125 VP8/9 a=fmtp:125 CIF=2;QCIF=2;SQCIF=2 a=rtpmap:106 H264/9 a=fmtp:106 profile-level-id=42e01e; packetization-mode=1; max-br=452; max-mbps=11880 a=rtpmap:121 MP4V-ES/9 a=fmtp:121 profile-level-id=3 a=rtpmap:103 H263-1998/9 a=fmtp:103 CIF=2;QCIF=2;SQCIF=2 when I make Audio call requests I dont have the video part but at receiver since two clients can make video call they have Asterisks adds the Video Part in request sent to receiver,I dont want that part added , how I can delete it ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Call type in dial plan
Here is the thing, my sip client can call the same. Extension once as audio and once as video, so I cannt turn off video supportat reciever, what I guess can be done is in extension.conf , there must be flag or something I can manipulate ... Sent from my iPhone On ٠٣/٠١/٢٠١٢, at ٨:١٩ ص, "virendra bhati" mailto:virbh...@gmail.com>> wrote: Which is means like if you are using sip 1234 then give the details of [1234] into that open thread and relevent extensions details too On Tue, Jan 3, 2012 at 11:30 AM, Faraj Khasib mailto:fkha...@iconnecths.com>> wrote: Which is?! What I am missing how to set dail plan in extension.conf to pass call type as its Not convert request to video Sent from my iPhone On ٠٣/٠١/٢٠١٢, at ٧:٢٩ ص, "virendra bhati" mailto:virbh...@gmail.com>> wrote: Hi, Please give you sip phone name and sip.conf and extensions.conf details which is using for that communication. And CLI output of asterisk is also required. On Tue, Jan 3, 2012 at 9:59 AM, Faraj Khasib mailto:fkha...@iconnecths.com>> wrote: I use asterisk 1.6, my clients are sip clients, I dail using audio call in my clients but the request is recieved at the other client as video call request since I am enabling video support for sip Sent from my iPhone On ٠٢/٠١/٢٠١٢, at ١١:٤٩ م, "Doug Lytle" mailto:supp...@drdos.info>> wrote: > > Faraj Khasib wrote: >> Please help, I have tried many things I cannt make it work, when I make an >> audio call it is converted by asterisk to video call request > > Not that I can help, since I don't do any video calling. > > But, if you don't give any information about your system (OS and > version, Asterisk version and what type of phone you are using), you're > not likely to get much of a response. > > Doug > > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve neither Liberty nor Safety." > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Call type in dial plan
Which is?! What I am missing how to set dail plan in extension.conf to pass call type as its Not convert request to video Sent from my iPhone On ٠٣/٠١/٢٠١٢, at ٧:٢٩ ص, "virendra bhati" mailto:virbh...@gmail.com>> wrote: Hi, Please give you sip phone name and sip.conf and extensions.conf details which is using for that communication. And CLI output of asterisk is also required. On Tue, Jan 3, 2012 at 9:59 AM, Faraj Khasib mailto:fkha...@iconnecths.com>> wrote: I use asterisk 1.6, my clients are sip clients, I dail using audio call in my clients but the request is recieved at the other client as video call request since I am enabling video support for sip Sent from my iPhone On ٠٢/٠١/٢٠١٢, at ١١:٤٩ م, "Doug Lytle" mailto:supp...@drdos.info>> wrote: > > Faraj Khasib wrote: >> Please help, I have tried many things I cannt make it work, when I make an >> audio call it is converted by asterisk to video call request > > Not that I can help, since I don't do any video calling. > > But, if you don't give any information about your system (OS and > version, Asterisk version and what type of phone you are using), you're > not likely to get much of a response. > > Doug > > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve neither Liberty nor Safety." > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Call type in dial plan
I use asterisk 1.6, my clients are sip clients, I dail using audio call in my clients but the request is recieved at the other client as video call request since I am enabling video support for sip Sent from my iPhone On ٠٢/٠١/٢٠١٢, at ١١:٤٩ م, "Doug Lytle" wrote: > > Faraj Khasib wrote: >> Please help, I have tried many things I cannt make it work, when I make an >> audio call it is converted by asterisk to video call request > > Not that I can help, since I don't do any video calling. > > But, if you don't give any information about your system (OS and > version, Asterisk version and what type of phone you are using), you're > not likely to get much of a response. > > Doug > > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve neither Liberty nor Safety." > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Call type in dial plan
Please help, I have tried many things I cannt make it work, when I make an audio call it is converted by asterisk to video call request, Please how to set the call type at extensions.conf, I tried setting the codec manually but didnt work also... any help .. any suggest will be great Thanx From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Monday, January 02, 2012 3:07 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Set Call type in dial plan Hi All, How to set C all type (Audio/Video) in dial plan? Regards Faraj Khasib -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Set Call type in dial plan
Hi All, How to set C all type (Audio/Video) in dial plan? Regards Faraj Khasib -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor Command Records separate channales
I attached log, but there is nothing unusual in it ...all normal ... From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] Sent: Wednesday, December 28, 2011 4:06 PM To: Faraj Khasib Subject: Your message to asterisk-users awaits moderator approval Your mail to 'asterisk-users' with the subject RE: [asterisk-users] Monitor Command Records separate channales Is being held until the list moderator can review it for approval. The reason it is being held: Message body is too big: 1004233 bytes with a limit of 40 KB Either the message will get posted to the list, or you will receive notification of the moderator's decision. If you would like to cancel this posting, please visit the following URL: http://lists.digium.com/mailman/confirm/asterisk-users/7c086c1398347b43db2e2984127934cd8cbde5c4 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor Command Records separate channales
It got stuck ... Sent from my iPhone On ٢٨/١٢/٢٠١١, at ١١:٢٩ م, "Danny Nicholas" wrote: > Try > # grep 'onitor' /var/log/asterisk/messages > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib > Sent: Wednesday, December 28, 2011 3:25 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Monitor Command Records separate channales > > but i tiried these commands and I didnt find anything about Monitor > [root@c-24-1-71-68 asterisk]# grep -R 'Monitor' * > [root@c-24-1-71-68 asterisk]# grep -R 'monitor' * > > From: asterisk-users-boun...@lists.digium.com > [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas > [da...@debsinc.com] > Sent: Wednesday, December 28, 2011 3:23 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: Re: [asterisk-users] Monitor Command Records separate channales > > I would wager that your setup dumps what would normally be in /v/l/a/full > into /v/l/a/messages > > -Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib > Sent: Wednesday, December 28, 2011 3:20 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Monitor Command Records separate channales > > see attached ... > > From: asterisk-users-boun...@lists.digium.com > [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas > [da...@debsinc.com] > Sent: Wednesday, December 28, 2011 3:18 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: Re: [asterisk-users] Monitor Command Records separate channales > > Even using Queue there should still be a /var/log/asterisk/full that records > the Monitor then the following Queue/Dial commands. What is in your > /var/log/asterisk? > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib > Sent: Wednesday, December 28, 2011 3:16 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Monitor Command Records separate channales > > My call happens with a queue , there is no full file but there is queue and > queue is useless, can u give me unix command to search all log files and > print moniter line? > > Sent from my iPhone > > On ٢٨/١٢/٢٠١١, at ١١:٠٩ م, "Danny Nicholas" wrote: > >> Asterisk -vvvrc >> Is how you would get it "live" >> After the fact you might find it in /var/log/asterisk/full >> >> >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj >> Khasib >> Sent: Wednesday, December 28, 2011 3:08 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Cc: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] Monitor Command Records separate >> channales >> >> Can u plz tell me how , I forgot how to run asterisk cli >> >> Sent from my iPhone >> >> On ٢٨/١٢/٢٠١١, at ١٠:٥٢ م, "Danny Nicholas" wrote: >> >>> Can you post a CLI output of the Monitor output? I'm supposing that >>> something in your $(STRFTIME) string might be "eating" the M option. >>> >>> -Original Message- >>> From: asterisk-users-boun...@lists.digium.com >>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj >>> Khasib >>> Sent: Wednesday, December 28, 2011 2:50 PM >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Cc: Asterisk Users Mailing List - Non-Commercial Discussion >>> Subject: Re: [asterisk-users] Monitor Command Records separate >>> channales >>> >>> Asterisk 1.6.2 but sox I don't know but now it is the latest version, >>> my problem is not mixing It's the same file but inside that file >>> two seperate records first callers then reciever >>> >>> Sent from my iPhone >>> >>> On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, "Danny Nicholas" wrote: >>> >>>> According to the monitor documentation, the format you specified >
Re: [asterisk-users] Monitor Command Records separate channales
but i tiried these commands and I didnt find anything about Monitor [root@c-24-1-71-68 asterisk]# grep -R 'Monitor' * [root@c-24-1-71-68 asterisk]# grep -R 'monitor' * From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, December 28, 2011 3:23 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Monitor Command Records separate channales I would wager that your setup dumps what would normally be in /v/l/a/full into /v/l/a/messages -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, December 28, 2011 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Monitor Command Records separate channales see attached ... From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, December 28, 2011 3:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Monitor Command Records separate channales Even using Queue there should still be a /var/log/asterisk/full that records the Monitor then the following Queue/Dial commands. What is in your /var/log/asterisk? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, December 28, 2011 3:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Monitor Command Records separate channales My call happens with a queue , there is no full file but there is queue and queue is useless, can u give me unix command to search all log files and print moniter line? Sent from my iPhone On ٢٨/١٢/٢٠١١, at ١١:٠٩ م, "Danny Nicholas" wrote: > Asterisk -vvvrc > Is how you would get it "live" > After the fact you might find it in /var/log/asterisk/full > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj > Khasib > Sent: Wednesday, December 28, 2011 3:08 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Monitor Command Records separate > channales > > Can u plz tell me how , I forgot how to run asterisk cli > > Sent from my iPhone > > On ٢٨/١٢/٢٠١١, at ١٠:٥٢ م, "Danny Nicholas" wrote: > >> Can you post a CLI output of the Monitor output? I'm supposing that >> something in your $(STRFTIME) string might be "eating" the M option. >> >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj >> Khasib >> Sent: Wednesday, December 28, 2011 2:50 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Cc: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] Monitor Command Records separate >> channales >> >> Asterisk 1.6.2 but sox I don't know but now it is the latest version, >> my problem is not mixing It's the same file but inside that file >> two seperate records first callers then reciever >> >> Sent from my iPhone >> >> On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, "Danny Nicholas" wrote: >> >>> According to the monitor documentation, the format you specified >>> should be calling SOX and mixing on call completion. What versions >>> of SOX and Asterisk are you using? >>> >>> -Original Message- >>> From: asterisk-users-boun...@lists.digium.com >>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj >>> Khasib >>> Sent: Wednesday, December 28, 2011 2:23 PM >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Subject: Re: [asterisk-users] Monitor Command Records separate >>> channales >>> >>> I installed SOX( it was not installed before). Will that solve my problem? >>> if not what are the parameter for the mixMonitor Command this is how >>> I use Monitor >>> exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y- >>> % >>> m >>> -%d_%H >>> :%M:%S)}_${SIP_HEADER(email)},m) >>> i
Re: [asterisk-users] Monitor Command Records separate channales
see attached ... From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, December 28, 2011 3:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Monitor Command Records separate channales Even using Queue there should still be a /var/log/asterisk/full that records the Monitor then the following Queue/Dial commands. What is in your /var/log/asterisk? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, December 28, 2011 3:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Monitor Command Records separate channales My call happens with a queue , there is no full file but there is queue and queue is useless, can u give me unix command to search all log files and print moniter line? Sent from my iPhone On ٢٨/١٢/٢٠١١, at ١١:٠٩ م, "Danny Nicholas" wrote: > Asterisk -vvvrc > Is how you would get it "live" > After the fact you might find it in /var/log/asterisk/full > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj > Khasib > Sent: Wednesday, December 28, 2011 3:08 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Monitor Command Records separate > channales > > Can u plz tell me how , I forgot how to run asterisk cli > > Sent from my iPhone > > On ٢٨/١٢/٢٠١١, at ١٠:٥٢ م, "Danny Nicholas" wrote: > >> Can you post a CLI output of the Monitor output? I'm supposing that >> something in your $(STRFTIME) string might be "eating" the M option. >> >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj >> Khasib >> Sent: Wednesday, December 28, 2011 2:50 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Cc: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] Monitor Command Records separate >> channales >> >> Asterisk 1.6.2 but sox I don't know but now it is the latest version, >> my problem is not mixing It's the same file but inside that file >> two seperate records first callers then reciever >> >> Sent from my iPhone >> >> On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, "Danny Nicholas" wrote: >> >>> According to the monitor documentation, the format you specified >>> should be calling SOX and mixing on call completion. What versions >>> of SOX and Asterisk are you using? >>> >>> -Original Message- >>> From: asterisk-users-boun...@lists.digium.com >>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj >>> Khasib >>> Sent: Wednesday, December 28, 2011 2:23 PM >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Subject: Re: [asterisk-users] Monitor Command Records separate >>> channales >>> >>> I installed SOX( it was not installed before). Will that solve my problem? >>> if not what are the parameter for the mixMonitor Command this is how >>> I use Monitor >>> exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y- >>> % >>> m >>> -%d_%H >>> :%M:%S)}_${SIP_HEADER(email)},m) >>> is Mix Monitor will have the same? >>> >>> From: asterisk-users-boun...@lists.digium.com >>> [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny >>> Nicholas [da...@debsinc.com] >>> Sent: Wednesday, December 28, 2011 2:20 PM >>> To: 'Asterisk Users Mailing List - Non-Commercial Discussion' >>> Subject: Re: [asterisk-users] Monitor Command Records separate >>> channales >>> >>> Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX. >>> >>> >>> -Original Message- >>> From: asterisk-users-boun...@lists.digium.com >>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj >>> Khasib >>> Sent: Wednesday, December 28, 2011 2:16 PM >>> To: asterisk-users@lists.digium.com >>> Subject: [asterisk-users] Monitor Command Records se
Re: [asterisk-users] Monitor Command Records separate channales
I already searched using grep for the monitor word ... It doesn't exists Sent from my iPhone On ٢٨/١٢/٢٠١١, at ١١:١٥ م, "Faraj Khasib" wrote: > My call happens with a queue , there is no full file but there is queue and > queue is useless, can u give me unix command to search all log files and > print moniter line? > > Sent from my iPhone > > On ٢٨/١٢/٢٠١١, at ١١:٠٩ م, "Danny Nicholas" wrote: > >> Asterisk -vvvrc >> Is how you would get it "live" >> After the fact you might find it in /var/log/asterisk/full >> >> >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib >> Sent: Wednesday, December 28, 2011 3:08 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Cc: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] Monitor Command Records separate channales >> >> Can u plz tell me how , I forgot how to run asterisk cli >> >> Sent from my iPhone >> >> On ٢٨/١٢/٢٠١١, at ١٠:٥٢ م, "Danny Nicholas" wrote: >> >>> Can you post a CLI output of the Monitor output? I'm supposing that >>> something in your $(STRFTIME) string might be "eating" the M option. >>> >>> -Original Message- >>> From: asterisk-users-boun...@lists.digium.com >>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj >>> Khasib >>> Sent: Wednesday, December 28, 2011 2:50 PM >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Cc: Asterisk Users Mailing List - Non-Commercial Discussion >>> Subject: Re: [asterisk-users] Monitor Command Records separate >>> channales >>> >>> Asterisk 1.6.2 but sox I don't know but now it is the latest version, >>> my problem is not mixing It's the same file but inside that file >>> two seperate records first callers then reciever >>> >>> Sent from my iPhone >>> >>> On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, "Danny Nicholas" wrote: >>> >>>> According to the monitor documentation, the format you specified >>>> should be calling SOX and mixing on call completion. What versions >>>> of SOX and Asterisk are you using? >>>> >>>> -Original Message- >>>> From: asterisk-users-boun...@lists.digium.com >>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj >>>> Khasib >>>> Sent: Wednesday, December 28, 2011 2:23 PM >>>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>>> Subject: Re: [asterisk-users] Monitor Command Records separate >>>> channales >>>> >>>> I installed SOX( it was not installed before). Will that solve my problem? >>>> if not what are the parameter for the mixMonitor Command this is how >>>> I use Monitor >>>> exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-% >>>> m >>>> -%d_%H >>>> :%M:%S)}_${SIP_HEADER(email)},m) >>>> is Mix Monitor will have the same? >>>> >>>> From: asterisk-users-boun...@lists.digium.com >>>> [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas >>>> [da...@debsinc.com] >>>> Sent: Wednesday, December 28, 2011 2:20 PM >>>> To: 'Asterisk Users Mailing List - Non-Commercial Discussion' >>>> Subject: Re: [asterisk-users] Monitor Command Records separate >>>> channales >>>> >>>> Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX. >>>> >>>> >>>> -Original Message- >>>> From: asterisk-users-boun...@lists.digium.com >>>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj >>>> Khasib >>>> Sent: Wednesday, December 28, 2011 2:16 PM >>>> To: asterisk-users@lists.digium.com >>>> Subject: [asterisk-users] Monitor Command Records separate channales >>>> >>>> Hi All, >>>> I am trying to record Call, but when the call is done I have one file >>>> but the conversation inside it is separate into calls conversation >>>> and receiver its single file but separate recording, How can I >>>> make it mixed together so the conversation will be normal? >>>&g
Re: [asterisk-users] Monitor Command Records separate channales
My call happens with a queue , there is no full file but there is queue and queue is useless, can u give me unix command to search all log files and print moniter line? Sent from my iPhone On ٢٨/١٢/٢٠١١, at ١١:٠٩ م, "Danny Nicholas" wrote: > Asterisk -vvvrc > Is how you would get it "live" > After the fact you might find it in /var/log/asterisk/full > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib > Sent: Wednesday, December 28, 2011 3:08 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Monitor Command Records separate channales > > Can u plz tell me how , I forgot how to run asterisk cli > > Sent from my iPhone > > On ٢٨/١٢/٢٠١١, at ١٠:٥٢ م, "Danny Nicholas" wrote: > >> Can you post a CLI output of the Monitor output? I'm supposing that >> something in your $(STRFTIME) string might be "eating" the M option. >> >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj >> Khasib >> Sent: Wednesday, December 28, 2011 2:50 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Cc: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] Monitor Command Records separate >> channales >> >> Asterisk 1.6.2 but sox I don't know but now it is the latest version, >> my problem is not mixing It's the same file but inside that file >> two seperate records first callers then reciever >> >> Sent from my iPhone >> >> On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, "Danny Nicholas" wrote: >> >>> According to the monitor documentation, the format you specified >>> should be calling SOX and mixing on call completion. What versions >>> of SOX and Asterisk are you using? >>> >>> -Original Message- >>> From: asterisk-users-boun...@lists.digium.com >>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj >>> Khasib >>> Sent: Wednesday, December 28, 2011 2:23 PM >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Subject: Re: [asterisk-users] Monitor Command Records separate >>> channales >>> >>> I installed SOX( it was not installed before). Will that solve my problem? >>> if not what are the parameter for the mixMonitor Command this is how >>> I use Monitor >>> exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-% >>> m >>> -%d_%H >>> :%M:%S)}_${SIP_HEADER(email)},m) >>> is Mix Monitor will have the same? >>> >>> From: asterisk-users-boun...@lists.digium.com >>> [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas >>> [da...@debsinc.com] >>> Sent: Wednesday, December 28, 2011 2:20 PM >>> To: 'Asterisk Users Mailing List - Non-Commercial Discussion' >>> Subject: Re: [asterisk-users] Monitor Command Records separate >>> channales >>> >>> Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX. >>> >>> >>> -Original Message- >>> From: asterisk-users-boun...@lists.digium.com >>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj >>> Khasib >>> Sent: Wednesday, December 28, 2011 2:16 PM >>> To: asterisk-users@lists.digium.com >>> Subject: [asterisk-users] Monitor Command Records separate channales >>> >>> Hi All, >>> I am trying to record Call, but when the call is done I have one file >>> but the conversation inside it is separate into calls conversation >>> and receiver its single file but separate recording, How can I >>> make it mixed together so the conversation will be normal? >>> Thanx >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>
Re: [asterisk-users] Monitor Command Records separate channales
Can u plz tell me how , I forgot how to run asterisk cli Sent from my iPhone On ٢٨/١٢/٢٠١١, at ١٠:٥٢ م, "Danny Nicholas" wrote: > Can you post a CLI output of the Monitor output? I'm supposing that > something in your $(STRFTIME) string might be "eating" the M option. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib > Sent: Wednesday, December 28, 2011 2:50 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Monitor Command Records separate channales > > Asterisk 1.6.2 but sox I don't know but now it is the latest version, my > problem is not mixing It's the same file but inside that file two > seperate records first callers then reciever > > Sent from my iPhone > > On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, "Danny Nicholas" wrote: > >> According to the monitor documentation, the format you specified >> should be calling SOX and mixing on call completion. What versions of >> SOX and Asterisk are you using? >> >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj >> Khasib >> Sent: Wednesday, December 28, 2011 2:23 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] Monitor Command Records separate >> channales >> >> I installed SOX( it was not installed before). Will that solve my problem? >> if not what are the parameter for the mixMonitor Command this is how I >> use Monitor >> exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m >> -%d_%H >> :%M:%S)}_${SIP_HEADER(email)},m) >> is Mix Monitor will have the same? >> >> From: asterisk-users-boun...@lists.digium.com >> [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas >> [da...@debsinc.com] >> Sent: Wednesday, December 28, 2011 2:20 PM >> To: 'Asterisk Users Mailing List - Non-Commercial Discussion' >> Subject: Re: [asterisk-users] Monitor Command Records separate >> channales >> >> Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX. >> >> >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj >> Khasib >> Sent: Wednesday, December 28, 2011 2:16 PM >> To: asterisk-users@lists.digium.com >> Subject: [asterisk-users] Monitor Command Records separate channales >> >> Hi All, >> I am trying to record Call, but when the call is done I have one file >> but the conversation inside it is separate into calls conversation and >> receiver its single file but separate recording, How can I make >> it mixed together so the conversation will be normal? >> Thanx >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory web
Re: [asterisk-users] Monitor Command Records separate channales
Asterisk 1.6.2 but sox I don't know but now it is the latest version, my problem is not mixing It's the same file but inside that file two seperate records first callers then reciever Sent from my iPhone On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, "Danny Nicholas" wrote: > According to the monitor documentation, the format you specified should be > calling SOX and mixing on call completion. What versions of SOX and > Asterisk are you using? > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib > Sent: Wednesday, December 28, 2011 2:23 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Monitor Command Records separate channales > > I installed SOX( it was not installed before). Will that solve my problem? > if not what are the parameter for the mixMonitor Command this is how I use > Monitor > exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H > :%M:%S)}_${SIP_HEADER(email)},m) > is Mix Monitor will have the same? > > From: asterisk-users-boun...@lists.digium.com > [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas > [da...@debsinc.com] > Sent: Wednesday, December 28, 2011 2:20 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: Re: [asterisk-users] Monitor Command Records separate channales > > Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX. > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib > Sent: Wednesday, December 28, 2011 2:16 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Monitor Command Records separate channales > > Hi All, > I am trying to record Call, but when the call is done I have one file but > the conversation inside it is separate into calls conversation and receiver > its single file but separate recording, How can I make it mixed > together so the conversation will be normal? > Thanx > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to > Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to > Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to > Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor Command Records separate channales
I installed SOX( it was not installed before). Will that solve my problem? if not what are the parameter for the mixMonitor Command this is how I use Monitor exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S)}_${SIP_HEADER(email)},m) is Mix Monitor will have the same? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, December 28, 2011 2:20 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Monitor Command Records separate channales Suggestion 1 - mixmonitor instead of monitor Suggestion 2 - SOX. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, December 28, 2011 2:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Monitor Command Records separate channales Hi All, I am trying to record Call, but when the call is done I have one file but the conversation inside it is separate into calls conversation and receiver its single file but separate recording, How can I make it mixed together so the conversation will be normal? Thanx -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitor Command Records separate channales
Hi All, I am trying to record Call, but when the call is done I have one file but the conversation inside it is separate into calls conversation and receiver its single file but separate recording, How can I make it mixed together so the conversation will be normal? Thanx -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Sip Media Call Type
Hi all, I am trying to make a SIP Video and Audio Call, Now when I add at the Asterisk the video Support and the right codec whether I make Audio or Video Call from my clients the Call will be received as Video Call, so the problem is if I make from one client Audio or Video Call it will be recieved as Video Call, Can you plz help me try to solve this problem? Where should I change the Call Media Type at Asterisks Regards Faraj Khasib -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
thank you for ur solution, I did this in dail plan yesterday ... it took me 5 hours to find that solution , I wish u replied to me earlier but thanx :) From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Torbjörn Abrahamsson [torbjorn.abrahams...@gmail.com] Sent: Monday, November 28, 2011 12:43 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message Well, It doesn't forward the INVITE at all, as asterisk is NOT a proxy. It creates a totally new INVITE when you issue the Dial application, with its own set of headers. Now, you can pass the Test header with something like this (taken from memory...): SipAddHeader(Test: ${SIP_HEADER(Test)}) Do that prior to the call to the Dial application, and you will see your header in the outgoing INVITE. Of course this means that your dial plan need to know which headers to pass. // T -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: den 28 november 2011 00:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message Any body knows how I can configure Asterisk SIP to pass all Header Parameters? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Sunday, November 27, 2011 4:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message thats my main question if u can see "Does Asterisk alter the Headersof INVITE Message" I am using ASterisk NOW proxy I didnt configure it to delete anything , Can u tell me how I can change it to pass that parameters? thanx From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov [abalas...@evaristesys.com] Sent: Sunday, November 27, 2011 4:41 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message On 11/27/2011 05:25 PM, Faraj Khasib wrote: > Yes, see attached ... Proxy server alter my "Test" custom header and > delete it, Is there a way to include it in message sent from SIP > Proxy to target? That would be a proxy configuration issue, wouldn't it? In principle, the proxy should be passing these messages through unmodified, unless you have an explicit configuration directive that instructs it to remove headers from the INVITE. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
Any body knows how I can configure Asterisk SIP to pass all Header Parameters? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Sunday, November 27, 2011 4:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message thats my main question if u can see "Does Asterisk alter the Headersof INVITE Message" I am using ASterisk NOW proxy I didnt configure it to delete anything , Can u tell me how I can change it to pass that parameters? thanx From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov [abalas...@evaristesys.com] Sent: Sunday, November 27, 2011 4:41 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message On 11/27/2011 05:25 PM, Faraj Khasib wrote: > Yes, see attached ... Proxy server alter my "Test" custom header and > delete it, Is there a way to include it in message sent from SIP > Proxy to target? That would be a proxy configuration issue, wouldn't it? In principle, the proxy should be passing these messages through unmodified, unless you have an explicit configuration directive that instructs it to remove headers from the INVITE. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
thats my main question if u can see "Does Asterisk alter the Headersof INVITE Message" I am using ASterisk NOW proxy I didnt configure it to delete anything , Can u tell me how I can change it to pass that parameters? thanx From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov [abalas...@evaristesys.com] Sent: Sunday, November 27, 2011 4:41 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message On 11/27/2011 05:25 PM, Faraj Khasib wrote: > Yes, see attached ... Proxy server alter my "Test" custom header and > delete it, Is there a way to include it in message sent from SIP > Proxy to target? That would be a proxy configuration issue, wouldn't it? In principle, the proxy should be passing these messages through unmodified, unless you have an explicit configuration directive that instructs it to remove headers from the INVITE. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
Yes, see attached ... Proxy server alter my "Test" custom header and delete it, Is there a way to include it in message sent from SIP Proxy to target? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov [abalas...@evaristesys.com] Sent: Sunday, November 27, 2011 4:19 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk alter the Headers ofINVITE Message On 11/27/2011 04:53 PM, Faraj Khasib wrote: > I tried that with my SIP Cleint but the custom Header is not reaching > the cleint ... Does the asketrisk delete that? Are you sure? Have you taken a packet capture to confirm? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Received (1128 bytes): 192.168.1.101:5060 <- 192.168.1.104:50495 INVITE sip:6500@192.168.1.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104:50495;branch=z9hG4bK1421826827;rport From: ;tag=194243250 To: Contact: ;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" Call-ID: 2447ed8f-b884-7731-6504-4626bbcb5511 CSeq: 947428168 INVITE Content-Type: application/sdp Content-Length: 257 Max-Forwards: 70 Route: Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel Test: testing Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER Privacy: none P-Access-Network-Info: ADSL;utran-cell-id-3gpp= User-Agent: Medcor Supported: 100rel v=0 o=doubango 1983 678901 IN IP4 192.168.1.104 s=- c=IN IP4 192.168.1.104 t=0 0 m=audio 38378 RTP/AVP 8 0 9 101 a=ptime:20 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:9 G722/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 Transaction(id='z9hG4bK1421826827' method=INVITE server=true) created. Sending (263 bytes): 192.168.1.101:5060 -> 192.168.1.104:50495 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.104:50495;branch=z9hG4bK1421826827;rport=50495 To: From: ;tag=194243250 Call-ID: 2447ed8f-b884-7731-6504-4626bbcb5511 CSeq: 947428168 INVITE Content-Length: 0 Transaction(id='z9hG4bK1421826827' method=INVITE server=true) Transaction timeout Timer started, will triger after 9. Transaction(id='z9hG4bK-d32f84d469d6407daffba73dccb7cadb' method=INVITE server=false) created. Transaction(id='z9hG4bK-d32f84d469d6407daffba73dccb7cadb' method=INVITE server=false) Timer A(requst retransmit timer) started, will triger after 500. Transaction(id='z9hG4bK-d32f84d469d6407daffba73dccb7cadb' method=INVITE server=false) Timer B(calling state timeout timer) started, will triger after 32000. Transaction(id='z9hG4bK-d32f84d469d6407daffba73dccb7cadb' method=INVITE server=false) Transcation timeout timer started,timeout after 18 ms Sending (1199 bytes): 192.168.1.101:5060 -> 192.168.1.101:59495 INVITE sip:6500@192.168.1.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.101;branch=z9hG4bK-d32f84d469d6407daffba73dccb7cadb;rport Via: SIP/2.0/UDP 192.168.1.104:50495;branch=z9hG4bK1421826827;rport=50495 From: ;tag=194243250 To: Contact: ;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" Call-ID: 2447ed8f-b884-7731-6504-4626bbcb5511 CSeq: 947428168 INVITE Content-Type: application/sdp Max-Forwards: 69 Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel Test: testing Allow: INVITE,ACK,CANCEL,BYE,MESSAGE,OPTIONS,NOTIFY,PRACK,UPDATE,REFER Privacy: none P-Access-Network-Info: ADSL;utran-cell-id-3gpp= User-Agent: Medcor Supported: 100rel Record-Route: Content-Length: 257 v=0 o=doubango 1983 678901 IN IP4 192.168.1.104 s=- c=IN IP4 192.168.1.104 t=0 0 m=audio 38378 RTP/AVP 8 0 9 101 a=ptime:20 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:9 G722/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 Received (421 bytes): 192.168.1.101:5060 <- 192.168.1.101:59495 SIP/2.0 100 Trying (sent from the Transaction Layer) Via: SIP/2.0/UDP 192.168.1.101;rport;branch=z9hG4bK-d32f84d469d6407daffba73dccb7cadb From: ;tag=194243250 To: Call-ID: 2447ed8f-b884-7731-6504-4626bbcb5511 CSeq: 947428168 INVITE Content-Length: 0 Via: SIP/2.0/UDP 192.168.1.104:50495;rport=50495;branch=z9
Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
I tried that with my SIP Cleint but the custom Header is not reaching the cleint ... Does the asketrisk delete that? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov [abalas...@evaristesys.com] Sent: Sunday, November 27, 2011 3:30 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message On 11/27/2011 04:27 PM, Faraj Khasib wrote: > Please guys anybody knows How can I send a unique token to the > Receiver at the Invite call? Is that possible? Custom SIP headers are a common way to do that. Try SIPAddHeader(). -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk alter the Headers of INVITE Message
Please guys anybody knows How can I send a unique token to the Receiver at the Invite call? Is that possible? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Sunday, November 27, 2011 11:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Does Asterisk alter the Headers of INVITE Message Hi all, I am trying to send an extra header in SIP INVITE Message , i.e (email="m...@me.com") but when I check the Message at the target that header is not there So I is Askterisk altering the Message and Is there away to include extra headers for SIP INVITE Message? Thank u -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does Asterisk alter the Headers of INVITE Message
Hi all, I am trying to send an extra header in SIP INVITE Message , i.e (email="m...@me.com") but when I check the Message at the target that header is not there So I is Askterisk altering the Message and Is there away to include extra headers for SIP INVITE Message? Thank u -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] I want to participate in development
Hi Asterisks Developers, I want to learn all about IP telephony and I was wondering If I can participate in development? Thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Asterisk Support SIP Video Call ?
Now I did, thank you for ur help and it works :D From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI [ad...@tootai.net] Sent: Wednesday, November 16, 2011 5:49 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk Support SIP Video Call ? Le 16/11/2011 10:23, Faraj Khasib a écrit : > Hi all, > I tried making a video SIP call using Asterisk But it didnt workonly > voice call works? Hi Faraj, Asterisk support H261, H263, H263+ and H264. Video calls are working since at least 1.4 version. You have to activate it by setting videosupport=yes in sip.conf -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does Asterisk Support SIP Video Call ?
Hi all, I tried making a video SIP call using Asterisk But it didnt workonly voice call works? Regards Faraj Khasib -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple SIP endpoint registrations
btw the call is one direction from clients to Call center My question can be rephrased can I make call without registration to an registered SIP account? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Tuesday, November 15, 2011 8:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multiple SIP endpoint registrations I have phone system and I am connecting Asterisk to it trunk. Now I want my iphone users (clients ) to call my call center which is in phone system by using the same SIP account the user will call asterik with for example 6000 as account then the asterik will forward the call via trunk to that Phone system. My question is this : Can all my iPhone users which are using the 6000 as an account call the call center ? with asterisk 1.7? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming [kpflem...@digium.com] Sent: Tuesday, November 15, 2011 8:25 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multiple SIP endpoint registrations On 11/15/2011 07:28 AM, Faraj Khasib wrote: > Hi guys, > I want to ask if its possible to make calls using one SIP account, > The problem is like this : I have an iPhone app and I want all my users to > call the same extension which is virtual extension to my call center, > so the iPhone app will be using the same SIP account for all users > lets say for example: > iPhone users uses 6000@mydomain to call 9000@my domain(which is the call > center) > Now My question is about the iPhone user part... Does the Asterisk 1.8 > support that all my iPhone users register with the same > account(6000@mydomain) and call that extension(dont worry about this > extension)? No Asterisk does not support multiple registrations to the same SIP account (AoR), but that is irrelevant in this case, because registrations are not used for placing calls *to* Asterisk, only receiving calls *from* Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple SIP endpoint registrations
I have phone system and I am connecting Asterisk to it trunk. Now I want my iphone users (clients ) to call my call center which is in phone system by using the same SIP account the user will call asterik with for example 6000 as account then the asterik will forward the call via trunk to that Phone system. My question is this : Can all my iPhone users which are using the 6000 as an account call the call center ? with asterisk 1.7? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming [kpflem...@digium.com] Sent: Tuesday, November 15, 2011 8:25 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multiple SIP endpoint registrations On 11/15/2011 07:28 AM, Faraj Khasib wrote: > Hi guys, > I want to ask if its possible to make calls using one SIP account, > The problem is like this : I have an iPhone app and I want all my users to > call the same extension which is virtual extension to my call center, > so the iPhone app will be using the same SIP account for all users > lets say for example: > iPhone users uses 6000@mydomain to call 9000@my domain(which is the call > center) > Now My question is about the iPhone user part... Does the Asterisk 1.8 > support that all my iPhone users register with the same > account(6000@mydomain) and call that extension(dont worry about this > extension)? No Asterisk does not support multiple registrations to the same SIP account (AoR), but that is irrelevant in this case, because registrations are not used for placing calls *to* Asterisk, only receiving calls *from* Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple SIP endpoint registrations
Hi guys, I want to ask if its possible to make calls using one SIP account, The problem is like this : I have an iPhone app and I want all my users to call the same extension which is virtual extension to my call center, so the iPhone app will be using the same SIP account for all users lets say for example: iPhone users uses 6000@mydomain to call 9000@my domain(which is the call center) Now My question is about the iPhone user part... Does the Asterisk 1.8 support that all my iPhone users register with the same account(6000@mydomain) and call that extension(dont worry about this extension)? Regards Faraj Khasib -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users