[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] 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
Re: [asterisk-users] Set Call type in dial plan
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
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
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 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" 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
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
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 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" 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 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" 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
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
Hi Might be it will help. Read it and set in extension as per your need. core show function CHANNEL -= Info about function 'CHANNEL' =- [Synopsis] Gets/sets various pieces of information about the channel. [Description] Gets/sets various pieces of information about the channel, additional may be available from the channel driver; see its documentation for details. Any requested that is not available on the current channel will return an empty string. [Syntax] CHANNEL(item) [Arguments] item Standard items (provided by all channel technologies) are: audioreadformat - R/O format currently being read. * audionativeformat - R/O format used natively for audio.* audiowriteformat - R/O format currently being written. callgroup - R/W call groups for call pickup. channeltype - R/O technology used for channel. language - R/W language for sounds played. musicclass - R/W class (from musiconhold.conf) for hold music. parkinglot - R/W parkinglot for parking. rxgain - R/W set rxgain level on channel drivers that support it. state - R/O state for channel tonezone - R/W zone for indications played transfercapability - R/W ISDN Transfer Capability, one of: SPEECH DIGITAL RESTRICTED_DIGITAL 3K1AUDIO DIGITAL_W_TONES VIDEO txgain - R/W set txgain level on channel drivers that support it. * videonativeformat - R/O format used natively for video* trace - R/W whether or not context tracing is enabled, only available *if CHANNEL_TRACE is defined*. *chan_sip* provides the following additional options: peerip - R/O Get the IP address of the peer. recvip - R/O Get the source IP address of the peer. from - R/O Get the URI from the From: header. uri - R/O Get the URI from the Contact: header. useragent - R/O Get the useragent. peername - R/O Get the name of the peer. t38passthrough - R/O '1' if T38 is offered or enabled in this channel, otherwise '0' rtpqos - R/O Get QOS information about the RTP stream This option takes two additional arguments: Argument 1: 'audio' Get data about the audio stream 'video' Get data about the video stream 'text' Get data about the text stream Argument 2: 'local_ssrc'Local SSRC (stream ID) 'local_lostpackets' Local lost packets 'local_jitter' Local calculated jitter 'local_maxjitter' Local calculated jitter (maximum) 'local_minjitter' Local calculated jitter (minimum) 'local_normdevjitter'Local calculated jitter (normal deviation) 'local_stdevjitter' Local calculated jitter (standard deviation) 'local_count' Number of received packets 'remote_ssrc' Remote SSRC (stream ID) 'remote_lostpackets'Remote lost packets 'remote_jitter' Remote reported jitter 'remote_maxjitter' Remote calculated jitter (maximum) 'remote_minjitter' Remote calculated jitter (minimum) 'remote_normdevjitter'Remote calculated jitter (normal deviation) 'remote_stdevjitter'Remote calculated jitter (standard deviation) 'remote_count' Number of transmitted packets 'rtt' Round trip time 'maxrtt'Round trip time (maximum) 'minrtt'Round trip time (minimum) 'normdevrtt'Round trip time (normal deviation) 'stdevrtt' Round trip time (standard deviation) 'all' All statistics (in a form suited to logging, but not for parsing) rtpdest - R/O Get remote RTP destination information. This option takes one additional argument: Argument 1: 'audio' Get audio destination 'video' Get video destination 'text' Get text destination *chan_iax2* provides the following additional options: peerip - R/O Get the peer's ip address. peername - R/O Get the peer's username. [See Also] Not available On Tue, Jan 3, 2012 at 11:53 AM, Faraj Khasib wrote: > 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" 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 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" wrot
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
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 wrote: > 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: ;transport=udp>;+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
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
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 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 [ > 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: ;transport=udp>;+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 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
