[asterisk-users] libpri Calling Line ID
Hey users, i've got a question about calling line id in libpri - zaptel with switchtype q.sig. My Q.Sig partner is a Siemens F900 (HiPoint). If I enable span debug i see messages from type CONNECT with some kind of bit field: Protocol Discriminator: Q.931 (8) len=87 Call Ref: len= 2 (reference 86/0x56) (Terminator) Message type: CONNECT (7) [1c 1d 9f aa 06 80 01 00 82 01 00 8b 01 00 a1 0f 02 02 4b 36 02 01 55 30 06 82 04 06 1c 08 40] Facility (len=31, codeset=0) [ 0x9f, 0xaa, 0x06, 0x80, 0x01, 0x00, 0x82, 0x01, 0x00, 0x8b, 0x01, 0x00, 0xa1, 0x0f, 0x02, 0x02, 'K6', 0x02, 0x01, 'U0', 0x06, 0x82, 0x04, 0x06, 0x1c, 0x08, 0x40 ] [1c 29 9f aa 06 80 01 00 82 01 00 8b 01 00 a1 1b 02 02 4b 45 02 01 02 a1 12 04 0d 4e 4f 52 44 4d 41 4e 4e 2c 45 52 49 43 02 01 01] Facility (len=43, codeset=0) [ 0x9f, 0xaa, 0x06, 0x80, 0x01, 0x00, 0x82, 0x01, 0x00, 0x8b, 0x01, 0x00, 0xa1, 0x1b, 0x02, 0x02, 'KE', 0x02, 0x01, 0x02, 0xa1, 0x12, 0x04, 0x0d, 'NORDMANN', 0x2c, 'ERIC', 0x02, 0x01, 0x01 ] [4c 06 00 80 32 35 37 37] Connected Number (len= 8) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Ext: 1 Presentation: Presentation permitted, user number not screened (0) '2577' ] There is also a name included: NORDMANN ERIC. Is there any way in asterisk to get this name in a variable or by any applikation command ? Thanks for your help in advance! Cheers, Michael Konietzny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfer app and DTMF via SIP info
Hello asterisk-users, I'm currently investigating a problem related to the Transfer app and DTMF tones via SipInfo. My setup depends on: Asterisk 1.2.10 Zaptel 1.2.8 libpri 1.2.3 Elmeg IP 290 (snom190) Wildcard TE400 (E1) The following dialplan is given: exten = 555, 1, Transfer(554); exten = 554, 1,Dial (SIP/tel3, 10, tT); exten = 554, 2,Dial (Zap/g1/017123123123, 10, tT); exten = 554, 3,Hangup(); If I dial 555 on my SIP phone it transfers to 554 and connecting me to that zap channel. Arriving there I'm not able to type ANY DTMF tones. If the Transfer is skipped the DTMF tones are available. I've included the SIP debugs to help you track the problem. Greetings and many thanks in advance, Michael Konietzny -- Executing Transfer(SIP/tel2-b721ef28, 554) in new stack Reliably Transmitting (no NAT) to 192.168.97.21:2054: SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 192.168.97.21:2054;branch=z9hG4bK-gu4c0f6c0cim;rport;received=192.168.97.21 From: tel2 sip:[EMAIL PROTECTED];tag=r7pzlq4bdy To: sip:[EMAIL PROTECTED];user=phone;tag=as21b6ba81 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: MMS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Transfer sip:[EMAIL PROTECTED] Content-Length: 0 ... -- Called tel3 -- SIP/tel3-082c99c8 is ringing SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.97.21:2054;branch=z9hG4bK-zqrwcwsdcly4;rport;received=192.168.97.21 From: tel2 sip:[EMAIL PROTECTED];tag=lt4rnm3do0 To: sip:[EMAIL PROTECTED];tag=as20294491 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: MMS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 235 v=0 o=root 31556 31556 IN IP4 192.168.97.11 s=session c=IN IP4 192.168.97.11 t=0 0 m=audio 18426 RTP/AVP 8 3 0 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 telephone-event/8000 a=fmtp:0 0-16 a=silenceSupp:off - - - - Via: SIP/2.0/UDP 192.168.97.21:2054;branch=z9hG4bK-zqrwcwsdcly4;rport From: tel2 sip:[EMAIL PROTECTED];tag=lt4rnm3do0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 2 CANCEL Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:2054;line=pisnle1m Content-Length: 0 ... -- Called g1/017123123123 We're at 192.168.97.11 port 18426 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (no NAT) to 192.168.97.21:2054: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.97.21:2054;branch=z9hG4bK-zqrwcwsdcly4;rport;received=192.168.97.21 From: tel2 sip:[EMAIL PROTECTED];tag=lt4rnm3do0 To: sip:[EMAIL PROTECTED];tag=as20294491 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: MMS Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 235 v=0 o=root 31556 31556 IN IP4 192.168.97.11 s=session c=IN IP4 192.168.97.11 t=0 0 m=audio 18426 RTP/AVP 8 3 0 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 telephone-event/8000 a=fmtp:0 0-16 a=silenceSupp:off - - - - -- Hungup 'Zap/1-1' INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.97.21:2054;branch=z9hG4bK-7zpvewacvy6j;rport From: tel2 sip:[EMAIL PROTECTED];tag=dtndk3lw7m To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:2054;line=pisnle1m P-Key-Flags: keys=3 User-Agent: snom190/3.60x Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Session-Expires: 3600 Content-Type: application/sdp Content-Length: 287 v=0 o=root 1728010931 1728010931 IN IP4 192.168.97.21 s=call c=IN IP4 192.168.97.21 t=0 0 m=audio 62868 RTP/AVP 8 0 3 9 18 4 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:3 gsm/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=ptime:20 a=sendrecv ... INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.97.21:2054;branch=z9hG4bK-qgjwhifwkg39;rport From: tel2 sip:[EMAIL PROTECTED];tag=dtndk3lw7m To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:2054;line=pisnle1m P-Key-Flags: keys=3 User-Agent: snom190/3.60x Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Session-Expires: 3600 Proxy-Authorization: Digest username=tel2,realm=asterisk,nonce=61364bf6,uri=sip:[EMAIL PROTECTED];user=phone,response=5140f1d5f042256b8daf901b18c603af,algorithm=md5 Content-Type: application/sdp Content-Length: 287 v=0 o=root 1728010931 1728010931 IN IP4 192.168.97.21 s=call c=IN IP4 192.168.97.21 t=0 0 m=audio 62868 RTP/AVP 8 0 3 9 18 4 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:3 gsm/8000 a=rtpmap:9 g722
[Asterisk-Users] Queue - Log if caller disconnects
Hello List, i'm wondering if there is any way to get a AGI executed if a caller disconnects while he is INSIDE the queue application. If so, i would like to log the call as missed. Hope someone can help. Greetings, Michael ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Queue NOT using RoundRobin ?!?
hey, a patch for linear mode is posted to bugs.digium.com already: http://bugs.digium.com/view.php?id=7279 greetings, Michael Aaron Paxson schrieb: If someone can point me in the right direction, I'll look into it. I'm not a C programmer, but I *should* be able to find my way. I'm looking at app_queue.c I see the strategies defined, but nothing about how they are used. Is app_queue.c the file that does the calling? - Original Message - *From:* Alessio Focardi mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Cc:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] *Sent:* Thursday, June 29, 2006 2:07 PM *Subject:* Re: [Asterisk-Users] Call Queue NOT using RoundRobin ?!? Will you (or anyone else) be able to code this proposed circular or linear (what sounds more appropriate?) strategy and submit it for inclusion in HEAD ? Should be pretty easy, unfortunately I have very few programming skills. Regards ! P.S. here is a snippet from the wiki, whatever it means ! :) roundrobin mode remembers the last agent it _started_ with for a new call, and starts with the next agent in the list. If you have three agents, the first call will go to agent 1-2-3, the next call will go to 2-3-1, the next call will go to 3-2-1, etc. rrmemory mode remembers the last agent it tried to _call_, regardless of who it started with, so that the next call will go the agent after the last one who answered. If you have three agents and the first call rings 1-2 (and is answered), then the next call will ring 3-1 (and is answered), then the next call will ring 2-3-1, etc. For the first call, if agent 2 answered it in roundrobin mode, they would still be the first agent for the next call, but rrmemory mode will move past them. On 6/29/06, *Aaron Paxson* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: The linear function helps me too. I've built an extensive multi-queue technical support system strategy. Based on the initial queue, ALL calls goes to Tier1 first. Then, if Tier1 does not get the call (on the phone/away from desk), Tier2 should get it, so on, and so forth. In Tier1, the primary helpdesk technician (like your receptionist idea) takes ALL calls (That's what they were hired for). However, others can help out, if the pri technician is on the phone. Here's my question: If roundrobin strategy remembers the last call made, and sends the next call to the next number (and this is by design), then why on earth was the RRMemory strategy created?? Thanks for your response, Alessio. ~~Aaron - Original Message - *From:* Alessio Focardi mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Cc:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] *Sent:* Thursday, June 29, 2006 1:31 PM *Subject:* Re: [Asterisk-Users] Call Queue NOT using RoundRobin ?!? Welcome to my personal hell ! :) I'have been discussing this previously on the list and also with some digium staff: to my experience there is NO way to archieve a linear distribution of calls from a queue. I mean When a call comes in first member of the queue is ring, then second, etc Subsequent calls take the same path: first, second and so on. Someone has suggested to use ringall with penalties (pretty esotic!) but also this is not working for the purpose. I was also told that nobody wants that (you insensitive clod!) even if this call distribution seems pretty logic in some case scenarios. (hint: a receptionist is first member of a queue and another person is the second ... receptionist goes for a pee and magically calls are rerouted to the backup operator after ringing to the first). Hope you can find out something to share, maybe we can also launch a count us initiative :) Alessio Focardi On 6/29/06, *Aaron Paxson* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have setup several Calling Queues, each setup with RoundRobin strategy. When I call the queue, the first member/agent phone rings. Great! I call it again, the second member/agent rings?? I thought that was
Re: [Asterisk-Users] Realtime queue_members and penalties nost escalating (clue anyone?)
Hey List, that issue could be interesting in that context: http://bugs.digium.com/view.php?id=7383 greetings, Michael Danny Froberg schrieb: Thanks for clearing that up Kevin. Now on to figure out how to PauseQueueMember when enough NOANSWER's has been detected so he don't fubar the entire queue. Would be alot cleaner than sending callers to ever higher level queues *sigh* Kevin P. Fleming wrote: Regardless of what strategy is used in the queues (roundrobin,rrmemory,ringall etc) it wont escalate on NOANSWER That is not how penalties are supposed to work. Calls are delivered to the lowest-penalty members that are considered available (i.e. not busy and not unreachable). The queue application does not turn 'noanswer' into 'unavailable'. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Zap/QSig with ChanIsAvailable
Hey, we're running an asterisk system connected to another telco system using qsig. I'm currently trying to use ChanIsAvailable to get the current phone status out of the foreign system. ChanIsAvailable always return 0 - UNKNOWN. The Qsig protocoll itself supports the feature to query the status of a given phone. Is there any other way beside to use ChanIsAvailable? Thanks in advance! Greetings, Michael ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trouble getting SMS working
Hey Mick, which version of asterisk are you using ? I've experienced problems after 1.2.8 with app_sms.so. It seams that the application is sending out an sms to the center. But this message was never recieved by my cellphone. To solve the problem i compiled app_sms.so from 1.2.7.1 source tree and moved it into the modules directory. Someone should contact a developer for that - since I experienced the problem I did not have time to track the problem down to it's origin. Hope this helps! Greetings, Michael Mick schrieb: I was thinking maybe the pap2 was doing 'something funny' so I tried turning off all unneeded services, fax, echo silence suppression etc. I also mucked about with the output gain, but nothing has made a difference so far. Yes I can hear the tones from * ! I will take the phone and try it on a fixed line and see if I can make it work there. Cheers Mick. 'sa' would appear to be the right option, as Asterisk in your case is answering the call as the message center (the phone is the 'terminal equipment') Would the pap2 be doing anything funny like waiting for fax tones or something before letting the tones go through? What happens if you just pick up the phone and dial the message center extension (199?)? Do you hear a (very) brief burst of tones? Tinker with sending very brief 'play' samples through and see how soon after you dial an extension you get the sounds. I have a pap2 but no SMS capable handset, but I might have a tinker James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Qsig
Hello Josué, the qsig feature came with Asterisk in version 1.2.x Greetings, Michael Josué Conti schrieb: Hello Michael, thank´s for help. But what´s version asterisk you use? The qsig protocol supported for what version? Best Regards Josué 2006/6/3, Michael Konietzny [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hello Josué, yes i currently only switched switchtype in zapata.conf to the value qsig. The only real PRI feature i've found out is the PRI_CAUSE variable set on Hangup(). Greetings, Michael Josué Conti schrieb: Michael, thank´s for this attention. I go to test with equipment Siemens HiPath and features. I sending for you and the list an email with the results of the tests, ok? How was zapata.conf of its asterisk with qsig? You it only changed switchtype=euroisdn, for switchtype=qsig? Thank you for its attention. Greetings Josué 2006/6/3, Michael Konietzny [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hello Josué, we're running Asterisk in combination of the T-Com Octopus E800 with QSig Protocoll. The protocoll itself is supported but some features are missing, or i didn't found out yet how to use them. I'm also interested in how to use qsig for determinating if other phones are available for calling and so on. Greetings, Michael Josué Conti schrieb: Hello all, as good? It would like to make a question, asterisk supports the protocol qsig, for interconnections in ISDN with equipment Siemens HiPath 4000 or same Ericsson MD110, so that it could identify to the name and the number of hosts and also to use some features of asterisk in the Siemens/Ericsson equipment. Greetings Josué ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com http://Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Qsig
Hello Josué, we're running Asterisk in combination of the T-Com Octopus E800 with QSig Protocoll. The protocoll itself is supported but some features are missing, or i didn't found out yet how to use them. I'm also interested in how to use qsig for determinating if other phones are available for calling and so on. Greetings, Michael Josué Conti schrieb: Hello all, as good? It would like to make a question, asterisk supports the protocol qsig, for interconnections in ISDN with equipment Siemens HiPath 4000 or same Ericsson MD110, so that it could identify to the name and the number of hosts and also to use some features of asterisk in the Siemens/Ericsson equipment. Greetings Josué ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mit freundlichen Grüßen Michael Konietzny e_mail: [EMAIL PROTECTED] handy: 0176 / 24 79 8656 phone: 03529 / 527597 address: Feldstrasse 7 01809 Heidenau ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Qsig
Hello Josué, yes i currently only switched switchtype in zapata.conf to the value qsig. The only real PRI feature i've found out is the PRI_CAUSE variable set on Hangup(). Greetings, Michael Josué Conti schrieb: Michael, thank´s for this attention. I go to test with equipment Siemens HiPath and features. I sending for you and the list an email with the results of the tests, ok? How was zapata.conf of its asterisk with qsig? You it only changed switchtype=euroisdn, for switchtype=qsig? Thank you for its attention. Greetings Josué 2006/6/3, Michael Konietzny [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hello Josué, we're running Asterisk in combination of the T-Com Octopus E800 with QSig Protocoll. The protocoll itself is supported but some features are missing, or i didn't found out yet how to use them. I'm also interested in how to use qsig for determinating if other phones are available for calling and so on. Greetings, Michael Josué Conti schrieb: Hello all, as good? It would like to make a question, asterisk supports the protocol qsig, for interconnections in ISDN with equipment Siemens HiPath 4000 or same Ericsson MD110, so that it could identify to the name and the number of hosts and also to use some features of asterisk in the Siemens/Ericsson equipment. Greetings Josué ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mit freundlichen Grüßen Michael Konietzny e_mail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] handy: 0176 / 24 79 8656 phone: 03529 / 527597 address: Feldstrasse 7 01809 Heidenau ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mit freundlichen Grüßen Michael Konietzny e_mail: [EMAIL PROTECTED] handy: 0176 / 24 79 8656 phone: 03529 / 527597 address: Feldstrasse 7 01809 Heidenau ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_queue and Real roundrobin
Hey guys, i'm wondering if there is any good way to get app_queue working in real roundrobin strategy. The idea is to specify a call list of, lets say, 3 agants. Those agents should always be called in the correct defined order. So all calls have to get the following agent priority: 1st Agent - 2nd Agent - 3rd Agent I've actually solved that by defining penelty for the accounts, but if the 1st Agent does not hear his/her phone and did not logged off correctly, the 2nd or 3rd agent has no chance to get the incoming call on his/her phone. It would be great if there is any solution - else it would be interesting how to send feature requests to asterisk-developers. Greetings from germany, Michael Konietzny ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_queue and Real roundrobin
Hello Kevin, thank you for your answer. Are you currently able to specify a date or period when the linar mode is implemented into app_queue ? Greetings, Michael Konietzny Kevin P. Fleming schrieb: Michael Konietzny wrote: i'm wondering if there is any good way to get app_queue working in real roundrobin strategy. The idea is to specify a call list of, lets say, 3 agants. Those agents should always be called in the correct defined order. So all calls have to get the following agent priority: 1st Agent - 2nd Agent - 3rd Agent This is not roundrobin, it's linear. We don't have a linear queue strategy at this time. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mit freundlichen Grüßen Michael Konietzny e_mail: [EMAIL PROTECTED] handy: 0176 / 24 79 8656 phone: 03529 / 527597 address: Feldstrasse 7 01809 Heidenau ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
