Re: [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug
i tried it and it wont work with rtcachefriend=yes On Fri, Feb 10, 2012 at 11:56 PM, JR Richardson wrote: > > I am facing an issue with Peer registration in my asterisk server . > > > > I am using asterisk version 1.8.5.0 and using SIP real-time > > architecture.when i am doing registration it registered fine on asterisk > > as peer is available in Database. > > > > But now i am doing 'sip reload' or 'reload' due to some reason my peer > > registration is going out and i cannot able to call that peer even though > > in SIP client it shows me 'registered'. > > > > Can any body elaborate on this issue which settings i need to put in > > sip.conf. > > > > I also tried to follow this patch > > https://issues.asterisk.org/view.php?id=14196 But it allready applied in > > code base so why it wont work? > > > > Here is my sip.conf settings. > > > > [general] > > context=from-internal; Default context for incoming cal > > rtcachefriends=no > > rtupdate=yes > > rtautoclear=yes > > rtsavesysname=yes > > callcounter = yes > > callevents=yes > > bindport=5060; UDP Port to bind to (SIP standard port is > 5060) > > srvlookup=yes; Enable DNS SRV lookups on outbound calls > > pedantic=yes; Enable slow, pedantic checking for Pingtel > > tos=184; Set IP QoS to either a keyword or numeric val > > tos_sip=cs3; Sets TOS for SIP packets. > > tos_audio=ef ; Sets TOS for RTP audio packets. > > tos=lowdelay; lowdelay,throughput,reliability,mincost,none > > maxexpiry=3600; Max length of incoming registration we allow > > defaultexpiry=120; Default length of incoming/outoing > registration > > preferred_codec_only=yes > > disallow=all; First disallow all codecs > > allow=ulaw; Allow codecs in order of preference > > allow=alaw > > insecure=invite > > language=en ; Default language setting for all > > users/peers > > rtpholdtimeout=300; Terminate call if 300 seconds of no RTP > > activity > > useragent=dhaval ; Allows you to change the user agent > string > > dtmfmode = rfc2833; Set default dtmfmode for sending DTMF. > Default: > > rfc2833 > > qualify=yes > > nat=yes > > ;canreinvite=yes > > directmedia=yes > > directrtpsetup=yes > > > > And here is DB fields snapshots. > > > > id: 1 > > name: 201 > > ipaddr: 172.18.100.243 > > port: 53624 > > regseconds: 1328716180 > > defaultuser: 201 > > fullcontact: NULL > >regserver: dhaval > >useragent: CSipSimple r1133 / b > > lastms: 554 > > host: dynamic > > type: friend > > context: from-internal > > permit: NULL > > deny: NULL > > secret: 201 > >md5secret: NULL > > remotesecret: NULL > >transport: NULL > > dtmfmode: NULL > > directmedia: yes > > nat: NULL > >allow: ulaw > > disallow: g729 > > insecure: invite > > callerid: NULL > > rfc2833compensate: NULL > > mailbox: NULL > > session-timers: NULL > > session-expires: NULL > >session-minse: NULL > > session-refresher: NULL > > > > Kindly help me to resolve this. > > > > Thanks > > Dhaval > > > > The first thing I would try is 'rtcachefriends=yes', that should do it. > > JR > -- > JR Richardson > Engineering for the Masses > > -- > _ > -- 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] Polycom firmware 4.0.1 and paging
In case anybody was following this thread, or someone Googles it in the future, here is the solution: This worked fine with Polycom firmware 3.3x: exten => s,n,SIPAddHeader(Alert-Info: ) For firmware 4.0+, apparently I needed to add info=, i.e.: exten => s,n,SIPAddHeader(Alert-Info: info=) Simple, yet quite obscure (for me at least). Mike > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of Mike > Sent: Monday, February 13, 2012 10:17 AM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging > > Thanks Dave, it at least gives me hope that my efforts aren`t wasted. > > Mike > > > -Original Message- > > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > > boun...@lists.digium.com] On Behalf Of Dave Fullerton > > Sent: Monday, February 13, 2012 9:39 AM > > To: asterisk-users@lists.digium.com > > Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging > > > > On 02/10/2012 05:30 PM, Mike wrote: > > > Hi, > > > > > > I just moved many Polycom phones from firmware v3 to 4.0.1b. > > > Anto-Answer simply stopped functioning. I can downgrade and make it > > > work, upgrading kills it again. There obviously is a difference in how > > > the newer firmware is treating this auto answer sip header. > > > > > > Can anybody tell me if they have Polycom firmware 4.x.x working with > > > auto-answer/paging? Just so I know it's worth my time to investigate, > > > as opposed to knowing it`s a Polycom firmware bug? If so, did you have > > > to make any changes to the SIP header sent to make Polycom phones auto > > answer? > > > > > > > I would second the others suggestions about rewriting the configs. > > Polycom made extensive changes between 3.2 and 3.3, and I think they > made > > a fair number of changes between 3.3 and 4.0. I have two phones that > I've > > upgraded to 4.0.1b for testing, a 550 and a spectralink 8440, and I > > believe I have auto answer working as you describe. Here's the pertinent > > snippet from my config: > > > > > > > > > > > voIpProt.SIP.alertInfo.1.class="ringAutoAnswer" > > voIpProt.SIP.alertInfo.1.value="intercom" > > voIpProt.SIP.alertInfo.2.class="ringAnswerMute" > > voIpProt.SIP.alertInfo.2.value="page" > > voIpProt.SIP.alertInfo.3.class="autoAnswer" > > voIpProt.SIP.alertInfo.3.value="silentanswer"> > > > > > > > > > > > > I have also added an section to adjust the ringer and timeouts > for > > these ring tones. > > > > -Dave > > > > -- > > _ > > -- 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] "conferenced" transfers
On Tue, Feb 14, 2012 at 3:10 PM, Andres wrote: > using the Cisco-Linksys SPA Phones you would: > 1) Receptionist Answers Call and hits 'Conf' button. > 2) Receptionist makes call and when answered hits 'Conf' again. > 3) Now everybody is talking > 4) Receptions hits 'Join' button. This releases the Receptionist from the > call and the other 2 parties are joined directly. I was just about to post exactly that. This is how we've taught our customers to do it, the feedback is positive. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] "conferenced" transfers
On the snom too Create a conferance and then press the transfer button. That will join the parties and release the receptionist -Original Message- From: Andres Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 14 Feb 2012 17:10:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Reply-To: and...@telesip.net, Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] "conferenced" transfers > > No, as I understand an attended transfer, there is no 3-way period where the > receptionist introduces the caller to someone else. In an attended transfer, > from the caller's perspective, he's talking to the receptionist, then he's on > hold, then he's talking to someone else. No different from a blind transfer, > from the caller perspective. > > using the Cisco-Linksys SPA Phones you would: 1) Receptionist Answers Call and hits 'Conf' button. 2) Receptionist makes call and when answered hits 'Conf' again. 3) Now everybody is talking 4) Receptions hits 'Join' button. This releases the Receptionist from the call and the other 2 parties are joined directly. -- Technical Support http://www.telesip.net -- _ -- 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] "conferenced" transfers
I think you can do the same thing with most Polycom phones. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres Sent: Tuesday, February 14, 2012 4:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] "conferenced" transfers > > No, as I understand an attended transfer, there is no 3-way period where the receptionist introduces the caller to someone else. In an attended transfer, from the caller's perspective, he's talking to the receptionist, then he's on hold, then he's talking to someone else. No different from a blind transfer, from the caller perspective. > > using the Cisco-Linksys SPA Phones you would: 1) Receptionist Answers Call and hits 'Conf' button. 2) Receptionist makes call and when answered hits 'Conf' again. 3) Now everybody is talking 4) Receptions hits 'Join' button. This releases the Receptionist from the call and the other 2 parties are joined directly. -- Technical Support http://www.telesip.net -- _ -- 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] "conferenced" transfers
No, as I understand an attended transfer, there is no 3-way period where the receptionist introduces the caller to someone else. In an attended transfer, from the caller's perspective, he's talking to the receptionist, then he's on hold, then he's talking to someone else. No different from a blind transfer, from the caller perspective. using the Cisco-Linksys SPA Phones you would: 1) Receptionist Answers Call and hits 'Conf' button. 2) Receptionist makes call and when answered hits 'Conf' again. 3) Now everybody is talking 4) Receptions hits 'Join' button. This releases the Receptionist from the call and the other 2 parties are joined directly. -- Technical Support http://www.telesip.net -- _ -- 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] Reading second rdnis
Hi, Does anyone how I could extract redirected number from a sip packet I have redirected a cell to a second cell which also rings a sip trunks and wish to route the call per rdnis The rdnis variable brings the first redirect (divert) which is the second cell but the first number also appears in the sip header as second divert Is there anyway I could easily extract the second divert header Asterisk 1.8 Thanks, Israel -- _ -- 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] Asterisk + Avaya (CM5.2) H.323 trunk Link
I am on a difinity system, Communication Manager version 5.2. Trying to use asterisk as my voice mail server and get rid of my Intuity Audix. On Tue, Feb 14, 2012 at 3:02 PM, Phil Frost wrote: > On Feb 14, 2012, at 14:56 , Dustin fails wrote: > > Anyone have an H.323 trunk tied between their Avaya and Asterisk box > that works? I am having some issues trying to get the two systems to > connect. I am using the ooh323 channel to try to make the connection > between the two system. I have all my configs if anyone would like to look > over them. If I do a trace on Avaya I get a denial event 1191: Network > Failure. > > > I have a trunk to an Avaya IP Office 500. It works mostly, but there are a > few lingering issues keeping me from using it in production. Namely, DTMF > seems to be passed between Asterisk and Avaya, but not from Asterisk, > through Avaya, to our SIP trunk. Also, I haven't yet gotten the caller ID > to be what I want. > > I can share more detail of my configs, but first, you didn't say what kind > of Avaya you have. IP Office? Or something else? > -- > Phil Frost > Macprofessionals > office 248-893-0738 > direct 248-662-0809 > > > > > -- > _ > -- 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] "conferenced" transfers
On Feb 14, 2012, at 15:34 , Danny Nicholas wrote: > As I read this, this is a regular "attended" transfer. No, as I understand an attended transfer, there is no 3-way period where the receptionist introduces the caller to someone else. In an attended transfer, from the caller's perspective, he's talking to the receptionist, then he's on hold, then he's talking to someone else. No different from a blind transfer, from the caller perspective. -- _ -- 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] "conferenced" transfers
As I read this, this is a regular "attended" transfer. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Frost Sent: Tuesday, February 14, 2012 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] "conferenced" transfers I'm wondering how one might implement a transfer where a receptionist introduces a caller to the recipient in a 3-way conference before hanging up, leaving the other two parties connected. Something like this, from the perspective of the customer: Customer: "Hi. I'd like to buy a widget." Receptionist: "Great. Let me connect you with someone in sales." (Customer on hold) Receptionist: "Hello customer. I have John here with me." John: "Hello." Receptionist: "John can sell you a widget. Have a great day." (Receptionist hangs up) (John and Customer continue the discussion) The problem is that in most systems I've seen, the 3-way is accomplished by the handset that initiates the conference mixing the two legs of the call. When that party (in this case, the receptionist) hangs up, the conference is over, and the other two parties are either disconnected or put on hold. I know there are ways to do server-side conferences - the challenge is making this no harder than a regular transfer, so the receptionist can do it comfortably. The usual model of dialing a number, entering a conference number, passcode, etc, is far too heavy for something as common as a transfer. In particular, I'm playing with a new Snom 870, and its drag-and-drop conference functionality is really great. However, I'm looking for any suggestions on how people skin this problem from a user interface perspective, and keep it friendly instead of frustrating for receptionists. -- _ -- 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] "conferenced" transfers
I'm wondering how one might implement a transfer where a receptionist introduces a caller to the recipient in a 3-way conference before hanging up, leaving the other two parties connected. Something like this, from the perspective of the customer: Customer: "Hi. I'd like to buy a widget." Receptionist: "Great. Let me connect you with someone in sales." (Customer on hold) Receptionist: "Hello customer. I have John here with me." John: "Hello." Receptionist: "John can sell you a widget. Have a great day." (Receptionist hangs up) (John and Customer continue the discussion) The problem is that in most systems I've seen, the 3-way is accomplished by the handset that initiates the conference mixing the two legs of the call. When that party (in this case, the receptionist) hangs up, the conference is over, and the other two parties are either disconnected or put on hold. I know there are ways to do server-side conferences - the challenge is making this no harder than a regular transfer, so the receptionist can do it comfortably. The usual model of dialing a number, entering a conference number, passcode, etc, is far too heavy for something as common as a transfer. In particular, I'm playing with a new Snom 870, and its drag-and-drop conference functionality is really great. However, I'm looking for any suggestions on how people skin this problem from a user interface perspective, and keep it friendly instead of frustrating for receptionists. -- _ -- 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] Asterisk + Avaya (CM5.2) H.323 trunk Link
On Feb 14, 2012, at 14:56 , Dustin fails wrote: > Anyone have an H.323 trunk tied between their Avaya and Asterisk box that > works? I am having some issues trying to get the two systems to connect. I am > using the ooh323 channel to try to make the connection between the two > system. I have all my configs if anyone would like to look over them. If I do > a trace on Avaya I get a denial event 1191: Network Failure. I have a trunk to an Avaya IP Office 500. It works mostly, but there are a few lingering issues keeping me from using it in production. Namely, DTMF seems to be passed between Asterisk and Avaya, but not from Asterisk, through Avaya, to our SIP trunk. Also, I haven't yet gotten the caller ID to be what I want. I can share more detail of my configs, but first, you didn't say what kind of Avaya you have. IP Office? Or something else? -- Phil Frost Macprofessionals office 248-893-0738 direct 248-662-0809 -- _ -- 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 + Avaya (CM5.2) H.323 trunk Link
Anyone have an H.323 trunk tied between their Avaya and Asterisk box that works? I am having some issues trying to get the two systems to connect. I am using the ooh323 channel to try to make the connection between the two system. I have all my configs if anyone would like to look over them. If I do a trace on Avaya I get a denial event 1191: Network Failure. 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
Re: [asterisk-users] Binding to 0.0.0.0 a security risk?
On 02/08/2012 04:29 AM, Tony Mountifield wrote: In article<4f324279.70...@message-id.plonk.de>, Jakob Hirsch wrote: Raj Mathur (राज माथ�र), 2012-02-08 03:27: Packets not going out on the same interface as the one they were received on is a general IP issue, not just for connectionless Right, this was a inaccuracy. It should say "Asterisk does not reply with the IP address with which packets were received". Asterisk (as most applications) does not care about network interfaces, it just handles IP addresses. protocols. The same behaviour can be seen with TCP too. Unless you mangle with iptables or something, all information about the received A tcp connection is defined by the tuple (source host&port, destination host&port), so if you write to a tcp socket, the kernel knows which source address it has to use (and also which destination address, so the application doesn't need to know that at all). As there's no such relation in udp, the application has to provide the destination address. The kernel then decides which source address to use, as long as the application did not bind() to a specific address. This is why some UDP servers such as for DNS and NTP create a separate socket bound specifically to each local IP address. Then by sending a response via the same socket as the request was received on, it can be reasonably sure that the response will go out on the right interface. Maybe Asterisk does or could do the same. I haven't checked. Well, 'Asterisk' is very broad, because really you are talking about each Asterisk module that can bind to sockets... and there are many of them. In the case of chan_iax2, multiple bindings are possible, and manual configuration could be done to individually bind to each address you want to provide services on (even if some of those addresses are configured on the same interface). Responses will be sent over the same socket the request was received on. In the case of chan_sip, only one UDP binding is possible (and one TCP/TLS binding). The code *could* be improved to handle multiple bindings, but it would be a large and invasive effort to do so. I've had thoughts in the past about this, and it would even possible to make this automatic (for systems where virtual hosting is being done), and have sockets automatically bound to new IP addresses that are discovered at run time... but that would still require that chan_sip be improved to properly handle fully multi-threaded operation for all of its data structures and operations. Alternatively, Olle Johannson has some patches that allow multiple instances of chan_sip to be loaded simultaneously; this could also be used to provide the sort of 'multiple binding' being talked about here. -- 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
Re: [asterisk-users] Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2
Hi, Am Dienstag, den 14.02.2012, 11:32 -0600 schrieb Kevin P. Fleming: > This does appear to be a bug in Asterisk; please open an issue in JIRA, > and post the issue number here, so we can get someone looking at this > ASAP. Thanks! > Done, issue ASTERISK-19358. If I can do anything to test something, let me know. Karsten -- _ -- 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] Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2
On 02/14/2012 11:19 AM, Karsten Wemheuer wrote:
Hi Kevin,
Am Dienstag, den 14.02.2012, 09:46 -0600 schrieb Kevin P. Fleming:
On 02/14/2012 09:30 AM, Karsten Wemheuer wrote:
Hi,
I got a problem with asterisk 1.8.9.2. The same scenario is working fine
in 1.8.8.2.
Asterisk calls a SIP phone via a proxy, proxy phone and asterisk are on
the same LAN, no NAT.
Asterisk sends the INVITE to the proxy, the proxy sends INVITE to the
phone. The phone sends 180 RINGING back to the proxy. The proxy sends
180 RINGING to asterisk. So far so good. If the calling side decides to
cancel the call, asterisk sends the CANCEL directly to the phone. The
phone doesn't find the call and answers 404. In asterisk 1.8.8.2
asterisk sends the CANCEL to the proxy, which sends the CANCEL to the
phone and all ist fine.
I think, the new behavior comes from the lines
parse_ok_contact(p, req);
if (!reinvite) {
build_route(p, req, 1);
}
which are inserted in the handling of provisional SIP response.
Am I doing something wrong or is this a bug?
It's impossible to answer that question without seeing the SIP
signaling. The answer will depend on what the proxy did to insert itself
in the path (or not) when it forwarded the 180 RINGING response to Asterisk.
I shorten the trace to (hopefully) the relevant things. Asterisk is on
192.168.10.72, port 25060, proxy is opnesips on the same machine with
port 5060, the phone which is ringing is on 192.168.10.221.
Asterisk => Proxy:
U 192.168.10.72:25060 -> 192.168.10.72:5060
INVITE sip:arthur@192.168.10.72 SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
Max-Forwards: 70.
From: "Max M..ller";tag=as3cafd135.
To:.
Contact:.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 INVITE.
... sdp cut of ...
Proxy => Asterisk
U 192.168.10.72:5060 -> 192.168.10.72:25060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
From: "Max M..ller";tag=as3cafd135.
To:.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 INVITE.
Proxy => phone
U 192.168.10.72:5060 -> 192.168.10.221:34381
INVITE sip:arthur@192.168.10.221:34381;line=478vzxb3 SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.72;branch=z9hG4bK24be.5163d992.0.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
Max-Forwards: 69.
From: "Max M..ller";tag=as3cafd135.
To:.
Contact:.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 INVITE.
... sdp cut of ...
Phone => Proxy
U 192.168.10.221:34381 -> 192.168.10.72:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.10.72;branch=z9hG4bK24be.5163d992.0.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
From: "Max M..ller";tag=as3cafd135.
To:;tag=cvovqkf6i5.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 INVITE.
Contact:;reg-id=1.
Proxy => Asterisk
U 192.168.10.72:5060 -> 192.168.10.72:25060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
From: "Max M..ller";tag=as3cafd135.
To:;tag=cvovqkf6i5.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 INVITE.
Contact:;reg-id=1.
When canceling the call, asterisk sends
Asterisk => Phone
U 192.168.10.72:25060 -> 192.168.10.221:34381
CANCEL sip:arthur@192.168.10.72 SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
Max-Forwards: 70.
From: "Max M..ller";tag=as3cafd135.
To:.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 CANCEL.
The Phone responds:
U 192.168.10.221:34381 -> 192.168.10.72:25060
SIP/2.0 404 Not found.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
From: "Max M..ller";tag=as3cafd135.
To:.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 CANCEL.
As noted in the earlier mail, this scenario is working in previous
versions (1,4.x up to asterisk 1.8.8.2).
Do You have any idea where the failure happens? Is it the proxy
configuration or is it at the asterisk side (maybe config or bug)?
This does appear to be a bug in Asterisk; please open an issue in JIRA,
and post the issue number here, so we can get someone looking at this
ASAP. Thanks!
--
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
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Re: [asterisk-users] Call holding with chan_capi
My apologies, I just realised I copied the wrong section of the debug log. So once again, when pressing the "park call" button, I get the following "capi debug" output: CAPI: ApplId=0x0002 Command=0x80 SubCommand=0x82 MsgNum=0xe446 NCCI=0x1403 FACILITY_IND ID=002 #0xe446 LEN=0018 Controller/PLCI/NCCI= 0x1403 FacilitySelector= 0x3 FacilityIndicationParameter = <02 80 00> -- ISDN_INTERN#02: unhandled FACILITY_IND supplementary function 8002 FACILITY_RESP ID=002 #0xe446 LEN=0015 Controller/PLCI/NCCI= 0x1403 FacilitySelector= 0x3 FacilityResponseParameters = default CAPI: ApplId=0x0002 Command=0x84 SubCommand=0x82 MsgNum=0xe447 NCCI=0x00011403 DISCONNECT_B3_IND ID=002 #0xe447 LEN=0015 Controller/PLCI/NCCI= 0x11403 Reason_B3 = 0x3301 NCPI= default DISCONNECT_B3_RESP ID=002 #0xe447 LEN=0012 Controller/PLCI/NCCI= 0x11403 -- _ -- 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] Call holding with chan_capi
In case this helps, when pressing the "Park Call" button, I get the following with "capi debug": DISCONNECT_REQ ID=002 #0x037e LEN=0013 Controller/PLCI/NCCI= 0x1303 AdditionalInfo = default CAPI: ApplId=0x0002 Command=0x04 SubCommand=0x81 MsgNum=0x037e NCCI=0x1303 DISCONNECT_CONFID=002 #0x037e LEN=0014 Controller/PLCI/NCCI= 0x1303 Info= 0x0 CAPI: ApplId=0x0002 Command=0x04 SubCommand=0x82 MsgNum=0xe3a5 NCCI=0x1303 DISCONNECT_IND ID=002 #0xe3a5 LEN=0014 Controller/PLCI/NCCI= 0x1303 Reason = 0x0 DISCONNECT_RESPID=002 #0xe3a5 LEN=0012 Controller/PLCI/NCCI= 0x1303 On 14/02/2012 18:18, Arik Raffael Funke wrote: Hi, I am using ISDN phones which have a "Park call" button. The idea is: you are on a call, push the button and hang up. You can then go to another phone and pickup the call without having to remember parking slots, etc. Unfortunately I cannot figure out how to get it to work with asterisk. I suspect it has something to do with capicommand(holdtype|local)... Does anybody use this isdn functionality with asterisk? Many thanks, Arik -- _ -- 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 implement outlook popup
2012/2/14, Luke Hamburg :
> Try TAPIRex
> http://www.tapirex.com/en/
>
> It's not free, but I've been using it with Asterisk + Outlook 2010
> successfully. Users can also click on the screenpop and it will open up the
> contact in Outlook. Pretty handy. You will need to make dialplan
> modifications to send out the call info to the user's workstations. TAPIRex
> implements a YAC listener on port 10629 so something like
>
> same => n,Set(cnam=CALLERID(name))
> same => n,Set(cnum=CALLERID(num))
> same => n,System(echo -e -n @CALL${cnam}~${cnum}|nc -w 1
> ${IP_of_screenpop_user} 10629)
>
> That code hasn't been tested -- it's just an example.
>
It's interesting : I'll give it close look !
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
> Sent: Tuesday, February 14, 2012 5:56 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] How to implement outlook popup
>
> Hi,
>
> For an RFP, I need to implement screen popup where caller names are searched
> in outlook folders.
> I would both consider free or paid solutions.
>
> Regards
>
> --
> _
> -- 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
>
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Re: [asterisk-users] Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2
Hi Kevin,
Am Dienstag, den 14.02.2012, 09:46 -0600 schrieb Kevin P. Fleming:
> On 02/14/2012 09:30 AM, Karsten Wemheuer wrote:
> > Hi,
> >
> > I got a problem with asterisk 1.8.9.2. The same scenario is working fine
> > in 1.8.8.2.
> >
> > Asterisk calls a SIP phone via a proxy, proxy phone and asterisk are on
> > the same LAN, no NAT.
> >
> > Asterisk sends the INVITE to the proxy, the proxy sends INVITE to the
> > phone. The phone sends 180 RINGING back to the proxy. The proxy sends
> > 180 RINGING to asterisk. So far so good. If the calling side decides to
> > cancel the call, asterisk sends the CANCEL directly to the phone. The
> > phone doesn't find the call and answers 404. In asterisk 1.8.8.2
> > asterisk sends the CANCEL to the proxy, which sends the CANCEL to the
> > phone and all ist fine.
> >
> > I think, the new behavior comes from the lines
> > parse_ok_contact(p, req);
> > if (!reinvite) {
> > build_route(p, req, 1);
> > }
> > which are inserted in the handling of provisional SIP response.
> >
> > Am I doing something wrong or is this a bug?
>
> It's impossible to answer that question without seeing the SIP
> signaling. The answer will depend on what the proxy did to insert itself
> in the path (or not) when it forwarded the 180 RINGING response to Asterisk.
>
I shorten the trace to (hopefully) the relevant things. Asterisk is on
192.168.10.72, port 25060, proxy is opnesips on the same machine with
port 5060, the phone which is ringing is on 192.168.10.221.
Asterisk => Proxy:
U 192.168.10.72:25060 -> 192.168.10.72:5060
INVITE sip:arthur@192.168.10.72 SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
Max-Forwards: 70.
From: "Max M..ller" ;tag=as3cafd135.
To: .
Contact: .
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 INVITE.
... sdp cut of ...
Proxy => Asterisk
U 192.168.10.72:5060 -> 192.168.10.72:25060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
From: "Max M..ller" ;tag=as3cafd135.
To: .
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 INVITE.
Proxy => phone
U 192.168.10.72:5060 -> 192.168.10.221:34381
INVITE sip:arthur@192.168.10.221:34381;line=478vzxb3 SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.72;branch=z9hG4bK24be.5163d992.0.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
Max-Forwards: 69.
From: "Max M..ller" ;tag=as3cafd135.
To: .
Contact: .
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 INVITE.
... sdp cut of ...
Phone => Proxy
U 192.168.10.221:34381 -> 192.168.10.72:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.10.72;branch=z9hG4bK24be.5163d992.0.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
From: "Max M..ller" ;tag=as3cafd135.
To: ;tag=cvovqkf6i5.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 INVITE.
Contact: ;reg-id=1.
Proxy => Asterisk
U 192.168.10.72:5060 -> 192.168.10.72:25060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
From: "Max M..ller" ;tag=as3cafd135.
To: ;tag=cvovqkf6i5.
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 INVITE.
Contact: ;reg-id=1.
When canceling the call, asterisk sends
Asterisk => Phone
U 192.168.10.72:25060 -> 192.168.10.221:34381
CANCEL sip:arthur@192.168.10.72 SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
Max-Forwards: 70.
From: "Max M..ller" ;tag=as3cafd135.
To: .
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 CANCEL.
The Phone responds:
U 192.168.10.221:34381 -> 192.168.10.72:25060
SIP/2.0 404 Not found.
Via: SIP/2.0/UDP 192.168.10.72:25060;branch=z9hG4bK6c6f4860.
From: "Max M..ller" ;tag=as3cafd135.
To: .
Call-ID: 4c640ebd11d192e15bfc6d3a667684ce@192.168.10.72.
CSeq: 102 CANCEL.
As noted in the earlier mail, this scenario is working in previous
versions (1,4.x up to asterisk 1.8.8.2).
Do You have any idea where the failure happens? Is it the proxy
configuration or is it at the asterisk side (maybe config or bug)?
Thanks,
Karsten
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[asterisk-users] Call holding with chan_capi
Hi, I am using ISDN phones which have a "Park call" button. The idea is: you are on a call, push the button and hang up. You can then go to another phone and pickup the call without having to remember parking slots, etc. Unfortunately I cannot figure out how to get it to work with asterisk. I suspect it has something to do with capicommand(holdtype|local)... Does anybody use this isdn functionality with asterisk? Many thanks, Arik -- _ -- 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] skip authentication for REGISTER
Thanks Kevin. Seems like remotesecret takes over if secret is not defined - I'll do further tests.. The authentication for REGISTERs and SUBSCRIBEs are done at a sip proxy (opensips) - I'll try to take care of the UAC authorization request for NOTIFY there (if possible). Regards, Matt > Date: Tue, 14 Feb 2012 09:44:38 -0600 > From: kpflem...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] skip authentication for REGISTER > > On 02/14/2012 08:43 AM, Matt Hamilton wrote: > > Hi, > > > > For REGISTER and SUBSCRIBE requests coming from UACs, is it possible to > > make Asterisk skip authentication even if a "secret" is defined in > > sip.conf for the peer; i.e. similar to insecure=invite for INVITE requests? > > > > If I leave "secret" blank, Asterisk doesn't require any authentication - > > this works as I want. However, I also use "SIP NOTIFY" to contact UACs > > (UACs are set to require authorization for NOTIFY), but without the > > "secret" defined, Asterisk can't send the correct authorization. > > You can use 'remotesecret' to set the secret string for Asterisk to use > to respond to authentication challenges. There isn't any way to make > REGISTER/SUBSCRIBE/etc. insecure like there is for INVITEs... I can't > imagine that many people would want unauthenticated REGISTERs to be > allowed :-) > > -- > 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
Re: [asterisk-users] Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2
On 02/14/2012 09:30 AM, Karsten Wemheuer wrote:
Hi,
I got a problem with asterisk 1.8.9.2. The same scenario is working fine
in 1.8.8.2.
Asterisk calls a SIP phone via a proxy, proxy phone and asterisk are on
the same LAN, no NAT.
Asterisk sends the INVITE to the proxy, the proxy sends INVITE to the
phone. The phone sends 180 RINGING back to the proxy. The proxy sends
180 RINGING to asterisk. So far so good. If the calling side decides to
cancel the call, asterisk sends the CANCEL directly to the phone. The
phone doesn't find the call and answers 404. In asterisk 1.8.8.2
asterisk sends the CANCEL to the proxy, which sends the CANCEL to the
phone and all ist fine.
I think, the new behavior comes from the lines
parse_ok_contact(p, req);
if (!reinvite) {
build_route(p, req, 1);
}
which are inserted in the handling of provisional SIP response.
Am I doing something wrong or is this a bug?
It's impossible to answer that question without seeing the SIP
signaling. The answer will depend on what the proxy did to insert itself
in the path (or not) when it forwarded the 180 RINGING response to 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
--
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Re: [asterisk-users] skip authentication for REGISTER
On 02/14/2012 08:43 AM, Matt Hamilton wrote: Hi, For REGISTER and SUBSCRIBE requests coming from UACs, is it possible to make Asterisk skip authentication even if a "secret" is defined in sip.conf for the peer; i.e. similar to insecure=invite for INVITE requests? If I leave "secret" blank, Asterisk doesn't require any authentication - this works as I want. However, I also use "SIP NOTIFY" to contact UACs (UACs are set to require authorization for NOTIFY), but without the "secret" defined, Asterisk can't send the correct authorization. You can use 'remotesecret' to set the secret string for Asterisk to use to respond to authentication challenges. There isn't any way to make REGISTER/SUBSCRIBE/etc. insecure like there is for INVITEs... I can't imagine that many people would want unauthenticated REGISTERs to be allowed :-) -- 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
[asterisk-users] Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2
Hi,
I got a problem with asterisk 1.8.9.2. The same scenario is working fine
in 1.8.8.2.
Asterisk calls a SIP phone via a proxy, proxy phone and asterisk are on
the same LAN, no NAT.
Asterisk sends the INVITE to the proxy, the proxy sends INVITE to the
phone. The phone sends 180 RINGING back to the proxy. The proxy sends
180 RINGING to asterisk. So far so good. If the calling side decides to
cancel the call, asterisk sends the CANCEL directly to the phone. The
phone doesn't find the call and answers 404. In asterisk 1.8.8.2
asterisk sends the CANCEL to the proxy, which sends the CANCEL to the
phone and all ist fine.
I think, the new behavior comes from the lines
parse_ok_contact(p, req);
if (!reinvite) {
build_route(p, req, 1);
}
which are inserted in the handling of provisional SIP response.
Am I doing something wrong or is this a bug?
Thanks,
Karsten
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[asterisk-users] skip authentication for REGISTER
Hi, For REGISTER and SUBSCRIBE requests coming from UACs, is it possible to make Asterisk skip authentication even if a "secret" is defined in sip.conf for the peer; i.e. similar to insecure=invite for INVITE requests? If I leave "secret" blank, Asterisk doesn't require any authentication - this works as I want. However, I also use "SIP NOTIFY" to contact UACs (UACs are set to require authorization for NOTIFY), but without the "secret" defined, Asterisk can't send the correct authorization. Thanks, Matt -- _ -- 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 implement outlook popup
Try TAPIRex
http://www.tapirex.com/en/
It's not free, but I've been using it with Asterisk + Outlook 2010
successfully. Users can also click on the screenpop and it will open up the
contact in Outlook. Pretty handy. You will need to make dialplan
modifications to send out the call info to the user's workstations. TAPIRex
implements a YAC listener on port 10629 so something like
same => n,Set(cnam=CALLERID(name))
same => n,Set(cnum=CALLERID(num))
same => n,System(echo -e -n @CALL${cnam}~${cnum}|nc -w 1
${IP_of_screenpop_user} 10629)
That code hasn't been tested -- it's just an example.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Tuesday, February 14, 2012 5:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to implement outlook popup
Hi,
For an RFP, I need to implement screen popup where caller names are searched
in outlook folders.
I would both consider free or paid solutions.
Regards
--
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Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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Re: [asterisk-users] Polycom IP331 Configuration
Thanks David. I will check it out. -Original message- From: "Klaverstyn, David C" To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Mon, Feb 13, 2012 04:34:30 GMT+00:00 Subject: Re: [asterisk-users] Polycom IP331 Configuration This may help you --> http://www.klaverstyn.com.au/david/wiki/index.php?title=Provision_Polycom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Johnson Sent: Monday, 13 February 2012 5:57 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom IP331 Configuration I hope this doesn't already exist, but I couldn't find anything to help. I am installing a brand new Asterisk server, and want to use the Polycom IP331 phones. Does anyone have any steps on how to configure these? I have softphones working just fine, but for some reason I can't find a clear step by step on provisioning the Polycoms. Any help is greatly appreciated! Mark J. -- _ -- 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] Asterisk 1.8.9.2 Now Available
We update from packages. Will this make its way to packages.asterisk.org or packages.digium.com? I double checked the sites. Thanks! EKG -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Development Team Sent: Thursday, February 09, 2012 4:43 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.8.9.2 Now Available The Asterisk Development Team has announced the release of Asterisk 1.8.9.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.9.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Fix SIP INFO DTMF handling for non-numeric codes --- (Closes issue ASTERISK-19290. Reported by: Ira Emus) * --- Fix crash in ParkAndAnnounce --- (Closes issue ASTERISK-19311. Reported-by: tootai) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.2 Thank you for your continued support of Asterisk! -- _ -- 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] chan_capi audio weirdness
Hi, I am trying to run asterisk on an AVM Fritz!Box Fon 7270 embedded DSL router. This works quite well after getting rid of the preinstalled phone server but I am encountering some unexpected behaviour. Background: I am using two CAPI controllers provided by the hardware - one in MSN mode for dialling out and - one in NT-mode, (DID) for the internal S0-Bus The problem is, I get no audio whatsoever until a channel is answered. Some of the symptoms of this are: - If I have an s-extension for the internal S0-Bus exten => s,1,Playtones(dial) I cannot hear the dialtone. It works however with: exten => s,1,Answer exten => s,n,Playtones(dial) - Similarly if I dial from internal to external with the extension: exten => _X.,1,Dial(CAPI/contr1/12345) I hear no progress indication. EVEN when using the r-option of the dial command. It works however with exten => _X.,1,Answer exten => _X.,n,Dial(CAPI/contr1/12345) Has anybody seen this before? Many thanks, Arik -- _ -- 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 associate agents - extensions?
Hi! I am setting up a little call center, but don't know how the agents system works, can you guys please give me a little help? I need to know how asterisk will know when I log agent X, and asterisk know that agent is in the IP Z with the extension Y. Thanks a lot. Hugs, ARPE -- _ -- 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 implement outlook popup
Hi, For an RFP, I need to implement screen popup where caller names are searched in outlook folders. I would both consider free or paid solutions. Regards -- _ -- 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] India Pune Pri call problem
Hi Virendra, I should have said, you can *set the callerid to one of the numbers allocated by them* for PRI, * and not to any other number*. Enjoy. --Satish Barot On Tue, Feb 14, 2012 at 1:31 PM, virendra bhati wrote: > Satish, > > As if I know, PRI provider give you PRI number at the time of purchase and > even billing documents will be made on the basis of the number only. So how > you can set another Caller-id number for that allotted number. > > But you can do only change the PRI number for outside world after > discussion with PRI provider. I did the same with Idea UP West circle. They > provide me 3 Callerid for single PRI lines for making OBD calls on that > circle. > > So all things is depends on PRI providers not at your end. > > > On Tue, Feb 14, 2012 at 1:24 PM, Satish Barot > wrote: > >> Indian Telcos do allow setting callerid on PRI line and you can set the >> callerid to one of the numbers allocated by them for PRI. >> >> --Satish Barot >> >> >> On Mon, Feb 13, 2012 at 6:49 PM, Ast Coder wrote: >> >>> India TRAI rules doesn't allow for CLID setting. They are backwards >>> minded. If you ever get them to do it let me know ;) >>> >>> -Bruce >>> >>> >>> On Mon, Feb 13, 2012 at 8:18 AM, Steven Howes >> > wrote: >>> On 13 Feb 2012, at 12:06, virendra bhati wrote: > You can't set callerid for outgoing calls in case of PRI. Why not? Every PRI I have used supported it. Is this a carrier-specific thing? S -- _ -- 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 >> > > > > -- > > Thanks and regards > > Virendra Bhati > +91-8885268942 > Software Engineer > E-mail-: virbh...@gmail.com > Skype id:- virbhati2 > > > -- > _ > -- 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] India Pune Pri call problem
Satish, As if I know, PRI provider give you PRI number at the time of purchase and even billing documents will be made on the basis of the number only. So how you can set another Caller-id number for that allotted number. But you can do only change the PRI number for outside world after discussion with PRI provider. I did the same with Idea UP West circle. They provide me 3 Callerid for single PRI lines for making OBD calls on that circle. So all things is depends on PRI providers not at your end. On Tue, Feb 14, 2012 at 1:24 PM, Satish Barot wrote: > Indian Telcos do allow setting callerid on PRI line and you can set the > callerid to one of the numbers allocated by them for PRI. > > --Satish Barot > > > On Mon, Feb 13, 2012 at 6:49 PM, Ast Coder wrote: > >> India TRAI rules doesn't allow for CLID setting. They are backwards >> minded. If you ever get them to do it let me know ;) >> >> -Bruce >> >> >> On Mon, Feb 13, 2012 at 8:18 AM, Steven Howes >> wrote: >> >>> On 13 Feb 2012, at 12:06, virendra bhati wrote: >>> > You can't set callerid for outgoing calls in case of PRI. >>> >>> Why not? Every PRI I have used supported it. Is this a carrier-specific >>> thing? >>> >>> S >>> -- >>> _ >>> -- 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 > -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- 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
