Re: [asterisk-users] Outbound Calls via Proxy to use Call ID from registration
On Mon, Aug 28, 2017, at 05:45 AM, Benoit Panizzon wrote: > Hello List > > > I work at an SIP Provider and we have added and SBC in front of our > > Voice Switch to protect it. > > Well using two peers for incomming and outgoing calls solve the > previous issue. > > Now I have a new one. > > The SBC in use needs to match incomming calls from the asterisk with > the call id used in the registration. > > We have tested this with a couple of PBX, which do use the call ID used > during registration automatically for outbound invites. > > Not so my asterisk server. > > So I assumed that when I refer to a 'peer' definition in the register > statement, I could make asterisk understand, that the registration and > outgoing peers belong together and then use the same call ID. Can you define what exactly you mean by call id? If you are referring to the Call-ID SIP header that's not how it works. It's unique within a dialog and not reused like that[1][2]. [1] https://tools.ietf.org/html/rfc3261#page-37 [2] https://tools.ietf.org/html/rfc3261#section-20.8 -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound calls
hi the issue still the same i have 2 trunks whe i configure the first in x-lite and the second in my server or my ip-phone snom320 directly from x-lite i can call my trunk without issue but when i try ti call from snom320 to x-lite or from my server asterisk using extension in x-lite the call all time is failed any help please thanks and regards 2015-03-20 19:28 GMT+00:00 Trey Hilyard kct...@gmail.com: So you are saying that it resolved the issue to activate voicemail on the device that sits past your trunk provider? That confuses me a little, but if your calls are working, that's great news. On Fri, Mar 20, 2015 at 1:44 PM Salaheddine Elharit salah.elharit...@gmail.com wrote: i noticed that when i active the voicemail in the IP-phone where the number 0033149xx is configured i can call this number without issue Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xx == Begin MixMonitor Recording SIP/101-010d -- SIP/FD-010e is making progress passing it to SIP/101-010d 0x2b393cfc2610 -- Probation passed - setting RTP source address to 192. 168.1.138:55542 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 -- SIP/FD-010e answered SIP/101-010d 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 thanks and 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 -- _ -- 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] outbound calls
thanks for your response i noticed that when i active the voicemail in the IP-phone where the number 0033149xx is configured i can call this number without issue the server asterisk and the ip-phone where the number is configured are in the same network 192.168.1.X Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xx == Begin MixMonitor Recording SIP/101-010d -- SIP/FD-010e is making progress passing it to SIP/101-010d 0x2b393cfc2610 -- Probation passed - setting RTP source address to 192. 168.1.138:55542 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 -- SIP/FD-010e answered SIP/101-010d 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 thanks and regards. 2015-03-20 18:39 GMT+00:00 Salaheddine Elharit salah.elharit...@gmail.com: thank you i noticed that when i active the voicemail in the IP-phone where the number 0033149xx is configured i can call this number without issue Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xx == Begin MixMonitor Recording SIP/101-010d -- SIP/FD-010e is making progress passing it to SIP/101-010d 0x2b393cfc2610 -- Probation passed - setting RTP source address to 192. 168.1.138:55542 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 -- SIP/FD-010e answered SIP/101-010d 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 thanks and regards. 2015-03-20 17:15 GMT+00:00 Trey Hilyard kct...@gmail.com: I am making some assumptions, but assuming the 217.195.xx.xxx is your provider, you are getting this back from them: Got SIP response 556 No address found back from 217.195.xx.xxx:5060 Are you sure that 0033149xx is the format the provider is expecting? You might try enabling SIP debug on the 217.195.xx.xx IP and seeing what the INVITE looks like, but normally a 556 indicates that your provider didn't have routing for either the R-URI or they didn't recognize that is was coming from you. You might compare the SIP INVITE coming from Asterisk to the one from Z-Lite and see where the differences are. On Fri, Mar 20, 2015 at 12:03 PM Salaheddine Elharit salah.elharit...@gmail.com wrote: hello list i have an issue related to outbound calls i can contact all the number except on number given by our provider in trunk the issue just when i configure my trunk in our server but when i configure the trunk directly in x-lite i can contact this number without issue below the cli == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0149xx@from-internal:1] Macro(SIP/101-0103, user-callerid,LIMIT,EXTERNAL,) in new stack -- Executing [s@macro-user-callerid:1] Set(SIP/101-0103, TOUCH_MONITOR=1426869820.301) in new stack -- Executing [s@macro-user-callerid:2] Set(SIP/101-0103, AMPUSER=101) in new stack -- Executing [s@macro-user-callerid:3] GotoIf(SIP/101-0103, 0?report) in new stack -- Executing [s@macro-user-callerid:4] ExecIf(SIP/101-0103, 1?Set(REALCALLERIDNUM=101)) in new stack -- Executing [s@macro-user-callerid:5] Set(SIP/101-0103, AMPUSER=101) in new stack -- Executing [s@macro-user-callerid:6] GotoIf(SIP/101-0103, 0?limit) in new stack -- Executing [s@macro-user-callerid:7] Set(SIP/101-0103, AMPUSERCIDNAME=101) in new stack -- Executing [s@macro-user-callerid:8] GotoIf(SIP/101-0103, 0?report) in new stack -- Executing [s@macro-user-callerid:9] Set(SIP/101-0103, AMPUSERCID=101) in new stack -- Executing [s@macro-user-callerid:10] Set(SIP/101-0103, __DIAL_OPTIONS=tr) in new stack -- Executing [s@macro-user-callerid:11] Set(SIP/101-0103, CALLERID(all)=101 101) in new stack -- Executing [s@macro-user-callerid:12] GotoIf(SIP/101-0103, 0?limit) in new stack -- Executing [s@macro-user-callerid:13] ExecIf(SIP/101-0103, 1?Set(GROUP(concurrency_limit)=101)) in new stack -- Executing [s@macro-user-callerid:14] ExecIf(SIP/101-0103, 0?Set(CHANNEL(language)=)) in new stack -- Executing [s@macro-user-callerid:15] GotoIf(SIP/101-0103, 1?continue) in new stack -- Goto (macro-user-callerid,s,28) -- Executing [s@macro-user-callerid:28] Set(SIP/101-0103, CALLERID(number)=101) in new stack -- Executing [s@macro-user-callerid:29] Set(SIP/101-0103, CALLERID(name)=101) in new stack -- Executing [s@macro-user-callerid:30] Set(SIP/101-0103, CDR(cnum)=101) in new stack -- Executing [s@macro-user-callerid:31] Set(SIP/101-0103, CDR(cnam)=101) in new stack -- Executing [s@macro-user-callerid:32] Set(SIP/101-0103, CHANNEL(language)=en) in new stack -- Executing
Re: [asterisk-users] outbound calls
i noticed that when i active the voicemail in the IP-phone where the number 0033149xx is configured i can call this number without issue Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xx == Begin MixMonitor Recording SIP/101-010d -- SIP/FD-010e is making progress passing it to SIP/101-010d 0x2b393cfc2610 -- Probation passed - setting RTP source address to 192. 168.1.138:55542 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 -- SIP/FD-010e answered SIP/101-010d 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 thanks and 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] outbound calls
I am making some assumptions, but assuming the 217.195.xx.xxx is your provider, you are getting this back from them: Got SIP response 556 No address found back from 217.195.xx.xxx:5060 Are you sure that 0033149xx is the format the provider is expecting? You might try enabling SIP debug on the 217.195.xx.xx IP and seeing what the INVITE looks like, but normally a 556 indicates that your provider didn't have routing for either the R-URI or they didn't recognize that is was coming from you. You might compare the SIP INVITE coming from Asterisk to the one from Z-Lite and see where the differences are. On Fri, Mar 20, 2015 at 12:03 PM Salaheddine Elharit salah.elharit...@gmail.com wrote: hello list i have an issue related to outbound calls i can contact all the number except on number given by our provider in trunk the issue just when i configure my trunk in our server but when i configure the trunk directly in x-lite i can contact this number without issue below the cli == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0149xx@from-internal:1] Macro(SIP/101-0103, user-callerid,LIMIT,EXTERNAL,) in new stack -- Executing [s@macro-user-callerid:1] Set(SIP/101-0103, TOUCH_MONITOR=1426869820.301) in new stack -- Executing [s@macro-user-callerid:2] Set(SIP/101-0103, AMPUSER=101) in new stack -- Executing [s@macro-user-callerid:3] GotoIf(SIP/101-0103, 0?report) in new stack -- Executing [s@macro-user-callerid:4] ExecIf(SIP/101-0103, 1?Set(REALCALLERIDNUM=101)) in new stack -- Executing [s@macro-user-callerid:5] Set(SIP/101-0103, AMPUSER=101) in new stack -- Executing [s@macro-user-callerid:6] GotoIf(SIP/101-0103, 0?limit) in new stack -- Executing [s@macro-user-callerid:7] Set(SIP/101-0103, AMPUSERCIDNAME=101) in new stack -- Executing [s@macro-user-callerid:8] GotoIf(SIP/101-0103, 0?report) in new stack -- Executing [s@macro-user-callerid:9] Set(SIP/101-0103, AMPUSERCID=101) in new stack -- Executing [s@macro-user-callerid:10] Set(SIP/101-0103, __DIAL_OPTIONS=tr) in new stack -- Executing [s@macro-user-callerid:11] Set(SIP/101-0103, CALLERID(all)=101 101) in new stack -- Executing [s@macro-user-callerid:12] GotoIf(SIP/101-0103, 0?limit) in new stack -- Executing [s@macro-user-callerid:13] ExecIf(SIP/101-0103, 1?Set(GROUP(concurrency_limit)=101)) in new stack -- Executing [s@macro-user-callerid:14] ExecIf(SIP/101-0103, 0?Set(CHANNEL(language)=)) in new stack -- Executing [s@macro-user-callerid:15] GotoIf(SIP/101-0103, 1?continue) in new stack -- Goto (macro-user-callerid,s,28) -- Executing [s@macro-user-callerid:28] Set(SIP/101-0103, CALLERID(number)=101) in new stack -- Executing [s@macro-user-callerid:29] Set(SIP/101-0103, CALLERID(name)=101) in new stack -- Executing [s@macro-user-callerid:30] Set(SIP/101-0103, CDR(cnum)=101) in new stack -- Executing [s@macro-user-callerid:31] Set(SIP/101-0103, CDR(cnam)=101) in new stack -- Executing [s@macro-user-callerid:32] Set(SIP/101-0103, CHANNEL(language)=en) in new stack -- Executing [0149xx@from-internal:2] Set(SIP/101-0103, MOHCLASS=default) in new stack -- Executing [0149xx@from-internal:3] Set(SIP/101-0103, _NODEST=) in new stack -- Executing [0149xx@from-internal:4] Gosub(SIP/101-0103, sub-record-check,s,1(out,0149xx,)) in new stack -- Executing [s@sub-record-check:1] Set(SIP/101-0103, REC_POLICY_MODE_SAVE=) in new stack -- Executing [s@sub-record-check:2] GotoIf(SIP/101-0103, 1?check) in new stack -- Goto (sub-record-check,s,7) -- Executing [s@sub-record-check:7] Set(SIP/101-0103, __MON_FMT=wav) in new stack -- Executing [s@sub-record-check:8] GotoIf(SIP/101-0103, 1?next) in new stack -- Goto (sub-record-check,s,11) -- Executing [s@sub-record-check:11] ExecIf(SIP/101-0103, 0?Return()) in new stack -- Executing [s@sub-record-check:12] ExecIf(SIP/101-0103, 0?Set(__REC_POLICY_MODE=)) in new stack -- Executing [s@sub-record-check:13] GotoIf(SIP/101-0103, 0?out,1) in new stack -- Executing [s@sub-record-check:14] Set(SIP/101-0103, __REC_STATUS=INITIALIZED) in new stack -- Executing [s@sub-record-check:15] Set(SIP/101-0103, NOW=1426869820) in new stack -- Executing [s@sub-record-check:16] Set(SIP/101-0103, __DAY=20) in new stack -- Executing [s@sub-record-check:17] Set(SIP/101-0103, __MONTH=03) in new stack -- Executing [s@sub-record-check:18] Set(SIP/101-0103, __YEAR=2015) in new stack -- Executing [s@sub-record-check:19] Set(SIP/101-0103, __TIMESTR=20150320-164340) in new stack -- Executing [s@sub-record-check:20] Set(SIP/101-0103, __FROMEXTEN=101) in new stack -- Executing [s@sub-record-check:21]
Re: [asterisk-users] outbound calls
So you are saying that it resolved the issue to activate voicemail on the device that sits past your trunk provider? That confuses me a little, but if your calls are working, that's great news. On Fri, Mar 20, 2015 at 1:44 PM Salaheddine Elharit salah.elharit...@gmail.com wrote: i noticed that when i active the voicemail in the IP-phone where the number 0033149xx is configured i can call this number without issue Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xx == Begin MixMonitor Recording SIP/101-010d -- SIP/FD-010e is making progress passing it to SIP/101-010d 0x2b393cfc2610 -- Probation passed - setting RTP source address to 192. 168.1.138:55542 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 -- SIP/FD-010e answered SIP/101-010d 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 thanks and 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
Re: [asterisk-users] outbound calls via google voice not answered by toll free numbers with ivrs
Apologize for following up to my own question, but wanted to mention that some toll free numbers with ivrs work fine. Only run into issues with certain numbers like the test number in my previous email. Any ideas? On Fri, May 13, 2011 at 10:26 AM, Gaurav P gaurav.lists+asterisk-us...@gmail.com wrote: Hi All, I'm using Asterisk 1.8.2 with outbound calls using Google Voice. I've been having issues calling several toll free numbers where the call 'is ringing' but never transitions to 'answered'. These are toll free numbers which are typically answered by an ivrs where you enter eg. a conference bridge number. I searched google and the closest reported issues I found are - https://issues.asterisk.org/view.php?id=18319 (on 1.6.x) and https://issues.asterisk.org/view.php?id=5266 (where the ibm support number listed does not work for my setup either) The number in the second ticket can be used as a test case - 800-426-7378- and I'm hoping someone has run into this before. I have already tried both 'auto' and 'rfc2833' settings for dtmfmode and can provide any additional details about my setup. Thanks in advance! -Gaurav -- _ -- 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] outbound calls not ringing still
3 sep 2009 kl. 00.27 skrev John A. Sullivan III: On Wed, 2009-09-02 at 21:31 +, Ott Rose wrote: i have posted this before but was unable to resolve it. i have some new info so i figured i would try again. the trace from bandwidth.com are below. they are telling me that the ip that is bold should be our ip not bandwidth.com. i have changed every setting that i can see and nothing fixes this. Where would i change this at? they cannot tell me. INVITE sip:+185993133...@216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP 216.82.224.202:5060;branch=z9hG4bK3691b08c;rport From:8592192438sip:8592192...@64.191.130.78;tag=as0707d433 To:sip:+185993133...@216.82.224.202 Contact:sip:8592192...@216.82.224.202 Call-ID: 0f3bdcc9171ef53148e7bab413aea...@64.191.130.78 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 02 Sep 2009 21:10:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 412 v=0 o=root 3831 3831 IN IP4 216.82.224.202 s=session c=IN IP4 216.82.224.202 t=0 0 m=audio 17050 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 12426 RTP/AVP 31 34 103 a=rtpmap:31 H261/9 a=rtpmap:34 H263/9 a=rtpmap:103 h263-1998/9 a=sendrecv snip I know very little about how ringing works but are they providing any kind of status information to you? Do you need to furnish the ring if they are not? It seems to me I saw quite a few articles about providing ring tone, what causes it to fail, and how to work around it. I assume you've searched for those already. Just a few thoughts - John It's very hard to say much without your configurations at hand. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound calls not ringing still
On Wed, 2009-09-02 at 21:31 +, Ott Rose wrote: i have posted this before but was unable to resolve it. i have some new info so i figured i would try again. the trace from bandwidth.com are below. they are telling me that the ip that is bold should be our ip not bandwidth.com. i have changed every setting that i can see and nothing fixes this. Where would i change this at? they cannot tell me. INVITE sip:+185993133...@216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP 216.82.224.202:5060;branch=z9hG4bK3691b08c;rport From:8592192438sip:8592192...@64.191.130.78;tag=as0707d433 To:sip:+185993133...@216.82.224.202 Contact:sip:8592192...@216.82.224.202 Call-ID: 0f3bdcc9171ef53148e7bab413aea...@64.191.130.78 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 02 Sep 2009 21:10:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 412 v=0 o=root 3831 3831 IN IP4 216.82.224.202 s=session c=IN IP4 216.82.224.202 t=0 0 m=audio 17050 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 12426 RTP/AVP 31 34 103 a=rtpmap:31 H261/9 a=rtpmap:34 H263/9 a=rtpmap:103 h263-1998/9 a=sendrecv snip I know very little about how ringing works but are they providing any kind of status information to you? Do you need to furnish the ring if they are not? It seems to me I saw quite a few articles about providing ring tone, what causes it to fail, and how to work around it. I assume you've searched for those already. Just a few thoughts - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound calls not ringing
Thanks that is very helpful info. I am still trying to figure out how asterisk and freepbx works together. what do I add in those files to get the ringing to work. I checked teh Dail options under General Options and its set to tr. Date: Thu, 20 Aug 2009 10:51:25 +1200 From: dun...@e-simple.co.nz To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] outbound calls not ringing Generally with FreePBX the ring options are set in the General Options - you can set the Dial options which are normally tr, but I guess that isn't working for you. The SIP files you could edit would have custom in their name, otherwise your changes will be overwritten when you reload freepbx You could put this in sip_general_custom.conf which will be included Cheers Duncan John A. Sullivan III wrote: Oops! - You're using FreePBX - someone who knows more about FreePBX will have to help you as I don't. May I also suggest that you bottom post in future responses rather than top post; that makes it a little easier to follow. Good luck - John On Wed, 2009-08-19 at 16:59 +, Ott Rose wrote: here is my sip.conf. i don't see it. ;; ; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ; ; this file must be done via the web gui. There are alternative files to make; ; custom modifications, details at: http://freepbx.org/configuration_files ; ;; ; [general] ; These files will all be included in the [general] context ; #include sip_general_additional.conf ;sip_general_custom.conf is the proper file location for placing any sip general ;options that you might need set. For example: enable and force the sip jitterbuffer. ;If these settings are desired they should be set the sip_general_custom.conf file. ; ; jbenable=yes ; jbforce=yes ; ;It is also the proper place to add the lines needed for sip nat'ing when going ;through a firewall. For nat'ing you'd need to add the following lines: ; nat=yes , externip= , localhost= , and optionally fromdomain= . ; #include sip_general_custom.conf ;sip_nat.conf is here for legacy support reasons and for those that upgrade ;from previous versions. If you have this file with lines in it please make ;sure they are not duplicated in sip_general_custom.conf, if so remove them ;from sip_nat.conf as sip_general_custom.conf will have precedence. #include sip_nat.conf ;sip_registrations_custom.conf is for any customizations you might need to do to ;the automatically generated registrations that FreePBX makes. ; #include sip_registrations_custom.conf #include sip_registrations.conf ; These files should all be expected to come after the [general] context ; #include sip_custom.conf #include sip_additional.conf ;sip_custom_post.conf If you have extra parameters that are needed for a ;extension to work to for example, those go here. So you have extension ;1000 defined in your system you start by creating a line [1000](+) in this ;file. Then on the next line add the extra parameter that is needed. ;When the sip.conf is loaded it will append your additions to the end of ;that extension. ; #include sip_custom_post.conf From: jsulli...@opensourcedevel.com To: asterisk-users@lists.digium.com Date: Wed, 19 Aug 2009 12:17:15 -0400 Subject: Re: [asterisk-users] outbound calls not ringing sip.conf On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote: we are using Aastra 57i i don't see that setting. where is it at? From: jsulli...@opensourcedevel.com To: asterisk-users@lists.digium.com Date: Wed, 19 Aug 2009 11:07:21 -0400 Subject: Re: [asterisk-users] outbound calls not ringing On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote: I put a post on here about my issues with outbound calls not ringing but i haven't resolved it. so i am trying again. When i dial any outside number i dont get a ring tone at all. when the person picks up and starts to talk i can hear them fine. it sounds great. How do I start to troubleshot this? snip What type of phones are giving you the problem? If I recall correctly, our SIP phones had this problem depending on how the destination handled signaling. We resolved it by adding progressinband=no (as opposed to the default never - at least I think it is the default) but this produces the problem of duplicate ring tones at times. Hope this helps - John -- John A. Sullivan III Open Source Development
Re: [asterisk-users] outbound calls not ringing
Have you tried putting a (,r) on your Dial command (dial dahdi/1/18005551212,60,r) ? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Wednesday, August 19, 2009 8:55 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] outbound calls not ringing I put a post on here about my issues with outbound calls not ringing but i haven't resolved it. so i am trying again. When i dial any outside number i dont get a ring tone at all. when the person picks up and starts to talk i can hear them fine. it sounds great. How do I start to troubleshot this? _ With Windows Live, you can organize, edit, and share your photos. Click http://www.windowslive.com/Desktop/PhotoGallery here. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound calls not ringing
On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote: I put a post on here about my issues with outbound calls not ringing but i haven't resolved it. so i am trying again. When i dial any outside number i dont get a ring tone at all. when the person picks up and starts to talk i can hear them fine. it sounds great. How do I start to troubleshot this? snip What type of phones are giving you the problem? If I recall correctly, our SIP phones had this problem depending on how the destination handled signaling. We resolved it by adding progressinband=no (as opposed to the default never - at least I think it is the default) but this produces the problem of duplicate ring tones at times. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound calls not ringing
we are using Aastra 57i i don't see that setting. where is it at? From: jsulli...@opensourcedevel.com To: asterisk-users@lists.digium.com Date: Wed, 19 Aug 2009 11:07:21 -0400 Subject: Re: [asterisk-users] outbound calls not ringing On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote: I put a post on here about my issues with outbound calls not ringing but i haven't resolved it. so i am trying again. When i dial any outside number i dont get a ring tone at all. when the person picks up and starts to talk i can hear them fine. it sounds great. How do I start to troubleshot this? snip What type of phones are giving you the problem? If I recall correctly, our SIP phones had this problem depending on how the destination handled signaling. We resolved it by adding progressinband=no (as opposed to the default never - at least I think it is the default) but this produces the problem of duplicate ring tones at times. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Hotmail® is up to 70% faster. Now good news travels really fast. http://windowslive.com/online/hotmail?ocid=PID23391::T:WLMTAGL:ON:WL:en-US:WM_HYGN_faster:082009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound calls not ringing
sip.conf On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote: we are using Aastra 57i i don't see that setting. where is it at? From: jsulli...@opensourcedevel.com To: asterisk-users@lists.digium.com Date: Wed, 19 Aug 2009 11:07:21 -0400 Subject: Re: [asterisk-users] outbound calls not ringing On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote: I put a post on here about my issues with outbound calls not ringing but i haven't resolved it. so i am trying again. When i dial any outside number i dont get a ring tone at all. when the person picks up and starts to talk i can hear them fine. it sounds great. How do I start to troubleshot this? snip What type of phones are giving you the problem? If I recall correctly, our SIP phones had this problem depending on how the destination handled signaling. We resolved it by adding progressinband=no (as opposed to the default never - at least I think it is the default) but this produces the problem of duplicate ring tones at times. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Hotmail® is up to 70% faster. Now good news travels really fast. Try it now. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound calls not ringing
here is my sip.conf. i don't see it. ;; ; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ; ; this file must be done via the web gui. There are alternative files to make ; ; custom modifications, details at: http://freepbx.org/configuration_files ; ;; ; [general] ; These files will all be included in the [general] context ; #include sip_general_additional.conf ;sip_general_custom.conf is the proper file location for placing any sip general ;options that you might need set. For example: enable and force the sip jitterbuffer. ;If these settings are desired they should be set the sip_general_custom.conf file. ; ; jbenable=yes ; jbforce=yes ; ;It is also the proper place to add the lines needed for sip nat'ing when going ;through a firewall. For nat'ing you'd need to add the following lines: ; nat=yes , externip= , localhost= , and optionally fromdomain= . ; #include sip_general_custom.conf ;sip_nat.conf is here for legacy support reasons and for those that upgrade ;from previous versions. If you have this file with lines in it please make ;sure they are not duplicated in sip_general_custom.conf, if so remove them ;from sip_nat.conf as sip_general_custom.conf will have precedence. #include sip_nat.conf ;sip_registrations_custom.conf is for any customizations you might need to do to ;the automatically generated registrations that FreePBX makes. ; #include sip_registrations_custom.conf #include sip_registrations.conf ; These files should all be expected to come after the [general] context ; #include sip_custom.conf #include sip_additional.conf ;sip_custom_post.conf If you have extra parameters that are needed for a ;extension to work to for example, those go here. So you have extension ;1000 defined in your system you start by creating a line [1000](+) in this ;file. Then on the next line add the extra parameter that is needed. ;When the sip.conf is loaded it will append your additions to the end of ;that extension. ; #include sip_custom_post.conf From: jsulli...@opensourcedevel.com To: asterisk-users@lists.digium.com Date: Wed, 19 Aug 2009 12:17:15 -0400 Subject: Re: [asterisk-users] outbound calls not ringing sip.conf On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote: we are using Aastra 57i i don't see that setting. where is it at? From: jsulli...@opensourcedevel.com To: asterisk-users@lists.digium.com Date: Wed, 19 Aug 2009 11:07:21 -0400 Subject: Re: [asterisk-users] outbound calls not ringing On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote: I put a post on here about my issues with outbound calls not ringing but i haven't resolved it. so i am trying again. When i dial any outside number i dont get a ring tone at all. when the person picks up and starts to talk i can hear them fine. it sounds great. How do I start to troubleshot this? snip What type of phones are giving you the problem? If I recall correctly, our SIP phones had this problem depending on how the destination handled signaling. We resolved it by adding progressinband=no (as opposed to the default never - at least I think it is the default) but this produces the problem of duplicate ring tones at times. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Hotmail® is up to 70% faster. Now good news travels really fast. Try it now. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users
Re: [asterisk-users] outbound calls not ringing
Oops! - You're using FreePBX - someone who knows more about FreePBX will have to help you as I don't. May I also suggest that you bottom post in future responses rather than top post; that makes it a little easier to follow. Good luck - John On Wed, 2009-08-19 at 16:59 +, Ott Rose wrote: here is my sip.conf. i don't see it. ;; ; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ; ; this file must be done via the web gui. There are alternative files to make; ; custom modifications, details at: http://freepbx.org/configuration_files ; ;; ; [general] ; These files will all be included in the [general] context ; #include sip_general_additional.conf ;sip_general_custom.conf is the proper file location for placing any sip general ;options that you might need set. For example: enable and force the sip jitterbuffer. ;If these settings are desired they should be set the sip_general_custom.conf file. ; ; jbenable=yes ; jbforce=yes ; ;It is also the proper place to add the lines needed for sip nat'ing when going ;through a firewall. For nat'ing you'd need to add the following lines: ; nat=yes , externip= , localhost= , and optionally fromdomain= . ; #include sip_general_custom.conf ;sip_nat.conf is here for legacy support reasons and for those that upgrade ;from previous versions. If you have this file with lines in it please make ;sure they are not duplicated in sip_general_custom.conf, if so remove them ;from sip_nat.conf as sip_general_custom.conf will have precedence. #include sip_nat.conf ;sip_registrations_custom.conf is for any customizations you might need to do to ;the automatically generated registrations that FreePBX makes. ; #include sip_registrations_custom.conf #include sip_registrations.conf ; These files should all be expected to come after the [general] context ; #include sip_custom.conf #include sip_additional.conf ;sip_custom_post.conf If you have extra parameters that are needed for a ;extension to work to for example, those go here. So you have extension ;1000 defined in your system you start by creating a line [1000](+) in this ;file. Then on the next line add the extra parameter that is needed. ;When the sip.conf is loaded it will append your additions to the end of ;that extension. ; #include sip_custom_post.conf From: jsulli...@opensourcedevel.com To: asterisk-users@lists.digium.com Date: Wed, 19 Aug 2009 12:17:15 -0400 Subject: Re: [asterisk-users] outbound calls not ringing sip.conf On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote: we are using Aastra 57i i don't see that setting. where is it at? From: jsulli...@opensourcedevel.com To: asterisk-users@lists.digium.com Date: Wed, 19 Aug 2009 11:07:21 -0400 Subject: Re: [asterisk-users] outbound calls not ringing On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote: I put a post on here about my issues with outbound calls not ringing but i haven't resolved it. so i am trying again. When i dial any outside number i dont get a ring tone at all. when the person picks up and starts to talk i can hear them fine. it sounds great. How do I start to troubleshot this? snip What type of phones are giving you the problem? If I recall correctly, our SIP phones had this problem depending on how the destination handled signaling. We resolved it by adding progressinband=no (as opposed to the default never - at least I think it is the default) but this produces the problem of duplicate ring tones at times. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Hotmail® is up to 70% faster. Now good news travels really fast. Try it now. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound calls not ringing
Generally with FreePBX the ring options are set in the General Options - you can set the Dial options which are normally tr, but I guess that isn't working for you. The SIP files you could edit would have custom in their name, otherwise your changes will be overwritten when you reload freepbx You could put this in sip_general_custom.conf which will be included Cheers Duncan John A. Sullivan III wrote: Oops! - You're using FreePBX - someone who knows more about FreePBX will have to help you as I don't. May I also suggest that you bottom post in future responses rather than top post; that makes it a little easier to follow. Good luck - John On Wed, 2009-08-19 at 16:59 +, Ott Rose wrote: here is my sip.conf. i don't see it. ;; ; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ; ; this file must be done via the web gui. There are alternative files to make; ; custom modifications, details at: http://freepbx.org/configuration_files ; ;; ; [general] ; These files will all be included in the [general] context ; #include sip_general_additional.conf ;sip_general_custom.conf is the proper file location for placing any sip general ;options that you might need set. For example: enable and force the sip jitterbuffer. ;If these settings are desired they should be set the sip_general_custom.conf file. ; ; jbenable=yes ; jbforce=yes ; ;It is also the proper place to add the lines needed for sip nat'ing when going ;through a firewall. For nat'ing you'd need to add the following lines: ; nat=yes , externip= , localhost= , and optionally fromdomain= . ; #include sip_general_custom.conf ;sip_nat.conf is here for legacy support reasons and for those that upgrade ;from previous versions. If you have this file with lines in it please make ;sure they are not duplicated in sip_general_custom.conf, if so remove them ;from sip_nat.conf as sip_general_custom.conf will have precedence. #include sip_nat.conf ;sip_registrations_custom.conf is for any customizations you might need to do to ;the automatically generated registrations that FreePBX makes. ; #include sip_registrations_custom.conf #include sip_registrations.conf ; These files should all be expected to come after the [general] context ; #include sip_custom.conf #include sip_additional.conf ;sip_custom_post.conf If you have extra parameters that are needed for a ;extension to work to for example, those go here. So you have extension ;1000 defined in your system you start by creating a line [1000](+) in this ;file. Then on the next line add the extra parameter that is needed. ;When the sip.conf is loaded it will append your additions to the end of ;that extension. ; #include sip_custom_post.conf From: jsulli...@opensourcedevel.com To: asterisk-users@lists.digium.com Date: Wed, 19 Aug 2009 12:17:15 -0400 Subject: Re: [asterisk-users] outbound calls not ringing sip.conf On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote: we are using Aastra 57i i don't see that setting. where is it at? From: jsulli...@opensourcedevel.com To: asterisk-users@lists.digium.com Date: Wed, 19 Aug 2009 11:07:21 -0400 Subject: Re: [asterisk-users] outbound calls not ringing On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote: I put a post on here about my issues with outbound calls not ringing but i haven't resolved it. so i am trying again. When i dial any outside number i dont get a ring tone at all. when the person picks up and starts to talk i can hear them fine. it sounds great. How do I start to troubleshot this? snip What type of phones are giving you the problem? If I recall correctly, our SIP phones had this problem depending on how the destination handled signaling. We resolved it by adding progressinband=no (as opposed to the default never - at least I think it is the default) but this produces the problem of duplicate ring tones at times. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound calls drop after 15 to 30 seconds.
On Mon, Aug 3, 2009 at 8:20 AM, Guillaume Yziquel guillaume.yziq...@citycable.ch wrote: Hello. I've set up and configured an Asterisk server to make SIP phone calls to external classic phones. However, it happens that after 15 or 30 seconds, the phone call drops. The SIP session still seems valid, but no sound comes through any more. How would you go through to troubleshoot this issue? All the best, Guillaume Yziquel. Make sure you have canreinvite set to no. Also, you may need to put an answer() in before your dial, I have dealt with that strangeness, call always drop at exactly 30 seconds. That solution worked for me, but I could see how it could mess up CDRs and billing for some applications. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound calls drop after 15 to 30 seconds.
Steve Totaro a écrit : On Mon, Aug 3, 2009 at 8:20 AM, Guillaume Yziquel guillaume.yziq...@citycable.ch wrote: Hello. I've set up and configured an Asterisk server to make SIP phone calls to external classic phones. However, it happens that after 15 or 30 seconds, the phone call drops. The SIP session still seems valid, but no sound comes through any more. How would you go through to troubleshoot this issue? All the best, Guillaume Yziquel. Make sure you have canreinvite set to no. It was already set to 'no' Also, you may need to put an answer() in before your dial, I have dealt with that strangeness, call always drop at exactly 30 seconds. Putting exten = _X.,n,Answer() in the dialplan doesn't change anything. That solution worked for me, but I could see how it could mess up CDRs and billing for some applications. Maybe I'm having a different issue than you've been experiencing. What's rather painful is that nothing appears to show in the Asterisk CLI when this happens since it's obviously not a problem with the SIP connection. How could I monitor the voice going in and out? All the best, Guillaume Yziquel. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound calls not reaching vitelity
On Tue, 2009-07-28 at 13:33 -0500, Tom Poe wrote: Any vitelity customers with pbxinaflash boxes? I'm able to call in-house, but failing to make outbound calls. My assigned server at vitelity is not reachable. I can ping to my ISP OK. Any help appreciated. Such as actually how to make email contact with support at vitelity. They're not responding. Thanks, Tom snip I'm not using pbxinaflash but I am using Vitelity and have had no problems at all - in fact very happy with them. They should have given you a management portal for your account probably portal.vitelity.net. In there, there is an option to open a trouble ticket. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound calls not reaching vitelity
John A. Sullivan III wrote: On Tue, 2009-07-28 at 13:33 -0500, Tom Poe wrote: Any vitelity customers with pbxinaflash boxes? I'm able to call in-house, but failing to make outbound calls. My assigned server at vitelity is not reachable. I can ping to my ISP OK. Any help appreciated. Such as actually how to make email contact with support at vitelity. They're not responding. Thanks, Tom snip I'm not using pbxinaflash but I am using Vitelity and have had no problems at all - in fact very happy with them. They should have given you a management portal for your account probably portal.vitelity.net. In there, there is an option to open a trouble ticket. Hope this helps - John Thanks, much. Will do. Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] outbound calls to sip urls
On Mon, Apr 24, 2006 at 04:46:29PM +0530, Ajit spake thusly: Hi, I wish to use the manager API to make an outbound call to a sip url,subsequently play a prompt and hangup.Any hints on how to acheive this/feasability will be much appreciated. I'm no expert, but it looks simple enough to me - just use the originate action to call with something like this: Action: Originate channel: SIP/[EMAIL PROTECTED] context: testcontext extension: extensiontosendtheprompt priority: 1 So that extension will just send the prompt and then hang up. -- Jon-o Addleman - http://redowl.dyndns.org ___ --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] Outbound calls are failing
Thanks
Inserting a w did resolve the problem. I saw another post from
today where somebody else is having the same problem with a
TDM2400P. Hopefully someday Asterisk will be coded to wait for a dial tone.
nb
On 4/19/06, Time Bandit [EMAIL PROTECTED] wrote:
When dialing an outbound number, sometimes all the digits are not dialed properly on the outside line. In the dial plan I added a SayDigits to the outbound rule and it properly reads back the entire number that was entered
on the phone before dialing.Is asterisk dialing too quickly, is there anyway to insert a pause or wait for a dial tone on the external line?* is probably starting to dial too fast. Try to add a w in your dial
string to make it wait.Like : Dial(ZAP/g0,w${EXTEN})w adds half a second pause. You can put more w to make it wait longer.hth___--Bandwidth and Colocation provided by
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Re: [Asterisk-Users] Outbound calls are failing
When dialing an outbound number, sometimes all the digits are not dialed
properly on the outside line. In the dial plan I added a SayDigits to the
outbound rule and it properly reads back the entire number that was entered
on the phone before dialing.
Is asterisk dialing too quickly, is there anyway to insert a pause or wait
for a dial tone on the external line?
* is probably starting to dial too fast. Try to add a w in your dial
string to make it wait.
Like : Dial(ZAP/g0,w${EXTEN})
w adds half a second pause. You can put more w to make it wait longer.
hth
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Re: [Asterisk-Users] Outbound calls through Broadvoice
Mike Raley wrote:
Hi all, a noob here, I am trying to get outbound calls through
asterisk working with Broadvoice.
I have consulted the following two online tutorials:
http://www.broadvoice.com/support_install_asterisk.html
http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice
in an effort to make outbound calls.
My current settings are as follows:
sip.conf
register =
[EMAIL PROTECTED]:SECRET:[EMAIL PROTECTED]/XX
where XX = our phone number including area code
and secret is our broadvoice defined secret
[sip.braodvoice.com]
maybe just a typo, br_OA_dvoice
I gone away from broadvoice, since they admitted to have troubles and I
had still to pay for NO phone call !!! (multiple lines)
bye
Ronald Wiplinger
type=peer
dynamic=yes
username=XX
fromuser=XX
authname=XX
user=phone
secret=SECRET
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
outboundproxy=sip.broadvoice.com
insecure=very
dtmfmode=inband
dtmf=inband
canreinvite=no
context=incoming
I receive the following error through asterisk when attempting a call:
Apr 8 13:08:43 WARNING[17425]: chan_sip.c:9634
handle_response_invite: Forbidden - wrong password on authentication
for INVITE to 'My Name sip:[EMAIL PROTECTED];tag=as23aa39db'
Now, we can receive incoming calls perfectly fine, but I just can't
wrap my head around what is wrong with the outgoing. I figure it's
got to be the way I am passing the phone number to call to Broadvoice:
exten = _3XNXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _3XNXXNXX,2,Congestion
or possibly, the fact that my name is showing up in the outbound call,
but the account isn't registered to my name, but someone else where I
work.
or my conf files are wrong somehow?
Otherwise, I got nothing.
Any help would be greatly appreciated by one frustrated noob!
oh, please CC me at mraley [at] syndiolemurgroup.com [remove the
mammal species]
Thanks!
Mike
--
Ronald Wiplinger (CEO of ELMIT)
http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com
Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208
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begin:vcard
fn:Ronald Wiplinger
n:Wiplinger;Ronald
org:ELMIT Co., Ltd.
adr:Shilin District;;5F., No.8, Alley 2, Lane 92, Dexing W. Road;Taipei;;11158;Taiwan
email;internet:[EMAIL PROTECTED]
title:CEO
tel;work:+886.2.2835.7765
tel;cell:+886.939.775.516
x-mozilla-html:TRUE
url:http://www.elmit.net
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Re: [Asterisk-Users] Outbound calls unpredictable
Frank wrote:
I've been looking through the archives and have not been able to find anyone
with a similar problem but perhaps I'm not searching in the right places. The
problem is that my outbound call sometimes go though and sometimes don't. If
someone can point me in the right direction it will be highly appreciated.
You could try altering you dial line so that it starts with a few w's:
exten = _9X.,1,Dial(Zap/1/www${EXTEN:1})
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] Outbound calls unpredictable
Thats amazing! Worked like a charm...any explanations as to why this happens?
On Wednesday 19 January 2005 03:21 am, Matt Riddell wrote:
Frank wrote:
I've been looking through the archives and have not been able to find
anyone with a similar problem but perhaps I'm not searching in the right
places. The problem is that my outbound call sometimes go though and
sometimes don't. If someone can point me in the right direction it will
be highly appreciated.
You could try altering you dial line so that it starts with a few w's:
exten = _9X.,1,Dial(Zap/1/www${EXTEN:1})
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Re: [Asterisk-Users] Outbound calls unpredictable
Frank wrote:
Thats amazing! Worked like a charm...any explanations as to why this happens?
Basically some connections require you to wait a little bit before
dialling the number. Without the w's it dials straight away. With them
it pauses and then dials.
On Wednesday 19 January 2005 03:21 am, Matt Riddell wrote:
Frank wrote:
I've been looking through the archives and have not been able to find
anyone with a similar problem but perhaps I'm not searching in the right
places. The problem is that my outbound call sometimes go though and
sometimes don't. If someone can point me in the right direction it will
be highly appreciated.
You could try altering you dial line so that it starts with a few w's:
exten = _9X.,1,Dial(Zap/1/www${EXTEN:1})
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] Outbound calls unpredictable
On Tue, 2005-01-18 at 05:20 -0500, Frank wrote: I've been looking through the archives and have not been able to find anyone with a similar problem but perhaps I'm not searching in the right places. The problem is that my outbound call sometimes go though and sometimes don't. If someone can point me in the right direction it will be highly appreciated. Care to give some logs or other information so we can think about helping you? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
