Re: [asterisk-users] Metasphere?
On 3/25/2010 8:13 AM, David Gibbons wrote: Hi All I'm involved in discussions with my carrier right now and am wondering if anyone has interconnected Asterisk to Metasphere via SIP? Yes, we're served by a Metaswitch usng SIP. Works fine. -Daryl -- _ -- 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] Problem building Asterisk 1.2.22
I'm having a problem building Asterisk 1.2.22. It fails in codecs/codec_zap.c on codec_zap.c is revision 62173. The OS is FC4. Here's the error. Can anyone help me with this? gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c -o codec_zap.o codec_zap.c codec_zap.c: In function ‘zap_framein’: codec_zap.c:143: error: dereferencing pointer to incomplete type codec_zap.c:145: error: dereferencing pointer to incomplete type codec_zap.c:147: error: dereferencing pointer to incomplete type codec_zap.c:147: error: dereferencing pointer to incomplete type codec_zap.c:152: error: dereferencing pointer to incomplete type codec_zap.c:152: error: dereferencing pointer to incomplete type codec_zap.c:152: error: dereferencing pointer to incomplete type codec_zap.c:154: error: dereferencing pointer to incomplete type codec_zap.c:154: error: dereferencing pointer to incomplete type codec_zap.c:154: error: dereferencing pointer to incomplete type codec_zap.c:154: error: dereferencing pointer to incomplete type codec_zap.c:155: error: dereferencing pointer to incomplete type codec_zap.c:158: error: dereferencing pointer to incomplete type codec_zap.c:158: error: dereferencing pointer to incomplete type codec_zap.c:158: error: dereferencing pointer to incomplete type codec_zap.c:159: error: dereferencing pointer to incomplete type codec_zap.c: In function ‘zap_frameout’: codec_zap.c:183: error: dereferencing pointer to incomplete type codec_zap.c:192: error: dereferencing pointer to incomplete type codec_zap.c:193: error: dereferencing pointer to incomplete type codec_zap.c:194: error: dereferencing pointer to incomplete type codec_zap.c:194: error: dereferencing pointer to incomplete type codec_zap.c:195: error: dereferencing pointer to incomplete type codec_zap.c:196: error: dereferencing pointer to incomplete type codec_zap.c:199: error: dereferencing pointer to incomplete type codec_zap.c:202: error: dereferencing pointer to incomplete type codec_zap.c:203: error: dereferencing pointer to incomplete type codec_zap.c:204: error: ‘ZT_TCOP_TRANSCODE’ undeclared (first use in this function) codec_zap.c:204: error: (Each undeclared identifier is reported only once codec_zap.c:204: error: for each function it appears in.) codec_zap.c:205: error: ‘ZT_TRANSCODE_OP’ undeclared (first use in this function) codec_zap.c: In function ‘zap_destroy’: codec_zap.c:219: error: ‘ZT_TCOP_RELEASE’ undeclared (first use in this function) codec_zap.c:220: error: ‘ZT_TRANSCODE_OP’ undeclared (first use in this function) codec_zap.c:223: error: dereferencing pointer to incomplete type codec_zap.c: In function ‘zap_new_alawtog723’: codec_zap.c:240: error: ‘ZT_TCOP_ALLOCATE’ undeclared (first use in this function) codec_zap.c:262: error: dereferencing pointer to incomplete type codec_zap.c:269: error: dereferencing pointer to incomplete type codec_zap.c:269: error: ‘ZT_TRANSCODE_MAGIC’ undeclared (first use in this function) codec_zap.c:270: error: dereferencing pointer to incomplete type codec_zap.c:271: error: dereferencing pointer to incomplete type codec_zap.c:277: error: dereferencing pointer to incomplete type codec_zap.c:278: error: dereferencing pointer to incomplete type codec_zap.c:279: error: ‘ZT_TRANSCODE_OP’ undeclared (first use in this function) codec_zap.c:281: error: dereferencing pointer to incomplete type codec_zap.c: In function ‘zap_new_ulawtog723’: codec_zap.c:297: error: ‘ZT_TCOP_ALLOCATE’ undeclared (first use in this function) codec_zap.c:319: error: dereferencing pointer to incomplete type codec_zap.c:326: error: dereferencing pointer to incomplete type codec_zap.c:326: error: ‘ZT_TRANSCODE_MAGIC’ undeclared (first use in this function) codec_zap.c:327: error: dereferencing pointer to incomplete type codec_zap.c:328: error: dereferencing pointer to incomplete type codec_zap.c:334: error: dereferencing pointer to incomplete type codec_zap.c:335: error: dereferencing pointer to incomplete type codec_zap.c:336: error: ‘ZT_TRANSCODE_OP’ undeclared (first use in this function) codec_zap.c:338: error: dereferencing pointer to incomplete type codec_zap.c: In function ‘zap_new_g723toalaw’: codec_zap.c:354: error: ‘ZT_TCOP_ALLOCATE’ undeclared (first use in this function) codec_zap.c:376: error: dereferencing pointer to incomplete type codec_zap.c:383: error: dereferencing pointer to incomplete type codec_zap.c:383: error: ‘ZT_TRANSCODE_MAGIC’ undeclared (first use in this function) codec_zap.c:384: error: dereferencing pointer to incomplete type codec_zap.c:385: error: dereferencing pointer to incomplete type codec_zap.c:391: error: dereferencing pointer to incomplete type codec_zap.c:392: error: dereferencing pointer to incomplete type codec_zap.c:393: error: ‘ZT_TRANSCODE_OP’ undeclared (first use in this function) codec_zap.c:395: error: dereferencing
Re: [asterisk-users] Problem building Asterisk 1.2.22
Correct. zaptel-1.2.12 is currently installed. I plan to install zaptel-1.2.19 as part of this upgrade. zaptel-1.2.19 compiled clean, but has not been installed yet. John covici wrote: I wonder what version of Zaptel you are using -- sounds like you have not installed a new version or you are using an older one. on Wednesday 07/18/2007 Daryl Jones([EMAIL PROTECTED]) wrote I'm having a problem building Asterisk 1.2.22. It fails in codecs/codec_zap.c on codec_zap.c is revision 62173. The OS is FC4. Here's the error. Can anyone help me with this? gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c -o codec_zap.o codec_zap.c codec_zap.c: In function ?zap_framein?: codec_zap.c:143: error: dereferencing pointer to incomplete type codec_zap.c:145: error: dereferencing pointer to incomplete type codec_zap.c:147: error: dereferencing pointer to incomplete type codec_zap.c:147: error: dereferencing pointer to incomplete type codec_zap.c:152: error: dereferencing pointer to incomplete type codec_zap.c:152: error: dereferencing pointer to incomplete type codec_zap.c:152: error: dereferencing pointer to incomplete type codec_zap.c:154: error: dereferencing pointer to incomplete type codec_zap.c:154: error: dereferencing pointer to incomplete type codec_zap.c:154: error: dereferencing pointer to incomplete type codec_zap.c:154: error: dereferencing pointer to incomplete type codec_zap.c:155: error: dereferencing pointer to incomplete type codec_zap.c:158: error: dereferencing pointer to incomplete type codec_zap.c:158: error: dereferencing pointer to incomplete type codec_zap.c:158: error: dereferencing pointer to incomplete type codec_zap.c:159: error: dereferencing pointer to incomplete type codec_zap.c: In function ?zap_frameout?: codec_zap.c:183: error: dereferencing pointer to incomplete type codec_zap.c:192: error: dereferencing pointer to incomplete type codec_zap.c:193: error: dereferencing pointer to incomplete type codec_zap.c:194: error: dereferencing pointer to incomplete type codec_zap.c:194: error: dereferencing pointer to incomplete type codec_zap.c:195: error: dereferencing pointer to incomplete type codec_zap.c:196: error: dereferencing pointer to incomplete type codec_zap.c:199: error: dereferencing pointer to incomplete type codec_zap.c:202: error: dereferencing pointer to incomplete type codec_zap.c:203: error: dereferencing pointer to incomplete type codec_zap.c:204: error: ?ZT_TCOP_TRANSCODE? undeclared (first use in this function) codec_zap.c:204: error: (Each undeclared identifier is reported only once codec_zap.c:204: error: for each function it appears in.) codec_zap.c:205: error: ?ZT_TRANSCODE_OP? undeclared (first use in this function) codec_zap.c: In function ?zap_destroy?: codec_zap.c:219: error: ?ZT_TCOP_RELEASE? undeclared (first use in this function) codec_zap.c:220: error: ?ZT_TRANSCODE_OP? undeclared (first use in this function) codec_zap.c:223: error: dereferencing pointer to incomplete type codec_zap.c: In function ?zap_new_alawtog723?: codec_zap.c:240: error: ?ZT_TCOP_ALLOCATE? undeclared (first use in this function) codec_zap.c:262: error: dereferencing pointer to incomplete type codec_zap.c:269: error: dereferencing pointer to incomplete type codec_zap.c:269: error: ?ZT_TRANSCODE_MAGIC? undeclared (first use in this function) codec_zap.c:270: error: dereferencing pointer to incomplete type codec_zap.c:271: error: dereferencing pointer to incomplete type codec_zap.c:277: error: dereferencing pointer to incomplete type codec_zap.c:278: error: dereferencing pointer to incomplete type codec_zap.c:279: error: ?ZT_TRANSCODE_OP? undeclared (first use in this function) codec_zap.c:281: error: dereferencing pointer to incomplete type codec_zap.c: In function ?zap_new_ulawtog723?: codec_zap.c:297: error: ?ZT_TCOP_ALLOCATE? undeclared (first use in this function) codec_zap.c:319: error: dereferencing pointer to incomplete type codec_zap.c:326: error: dereferencing pointer to incomplete type codec_zap.c:326: error: ?ZT_TRANSCODE_MAGIC? undeclared (first use in this function) codec_zap.c:327: error: dereferencing pointer to incomplete type codec_zap.c:328: error: dereferencing pointer to incomplete type codec_zap.c:334: error: dereferencing pointer to incomplete type codec_zap.c:335: error: dereferencing pointer to incomplete type codec_zap.c:336: error: ?ZT_TRANSCODE_OP? undeclared (first use in this function) codec_zap.c:338: error: dereferencing pointer to incomplete type codec_zap.c: In function ?zap_new_g723toalaw?: codec_zap.c:354: error: ?ZT_TCOP_ALLOCATE? undeclared (first
Re: [asterisk-users] Problem building Asterisk 1.2.22
That's what I needed to know. Thanks! John covici wrote: But asterisk will not compile till you install the correct version of zaptel. on Wednesday 07/18/2007 Daryl Jones([EMAIL PROTECTED]) wrote Correct. zaptel-1.2.12 is currently installed. I plan to install zaptel-1.2.19 as part of this upgrade. zaptel-1.2.19 compiled clean, but has not been installed yet. John covici wrote: I wonder what version of Zaptel you are using -- sounds like you have not installed a new version or you are using an older one. on Wednesday 07/18/2007 Daryl Jones([EMAIL PROTECTED]) wrote I'm having a problem building Asterisk 1.2.22. It fails in codecs/codec_zap.c on codec_zap.c is revision 62173. The OS is FC4. Here's the error. Can anyone help me with this? gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c -o codec_zap.o codec_zap.c codec_zap.c: In function ?zap_framein?: codec_zap.c:143: error: dereferencing pointer to incomplete type codec_zap.c:145: error: dereferencing pointer to incomplete type codec_zap.c:147: error: dereferencing pointer to incomplete type codec_zap.c:147: error: dereferencing pointer to incomplete type codec_zap.c:152: error: dereferencing pointer to incomplete type codec_zap.c:152: error: dereferencing pointer to incomplete type codec_zap.c:152: error: dereferencing pointer to incomplete type codec_zap.c:154: error: dereferencing pointer to incomplete type codec_zap.c:154: error: dereferencing pointer to incomplete type codec_zap.c:154: error: dereferencing pointer to incomplete type codec_zap.c:154: error: dereferencing pointer to incomplete type codec_zap.c:155: error: dereferencing pointer to incomplete type codec_zap.c:158: error: dereferencing pointer to incomplete type codec_zap.c:158: error: dereferencing pointer to incomplete type codec_zap.c:158: error: dereferencing pointer to incomplete type codec_zap.c:159: error: dereferencing pointer to incomplete type codec_zap.c: In function ?zap_frameout?: codec_zap.c:183: error: dereferencing pointer to incomplete type codec_zap.c:192: error: dereferencing pointer to incomplete type codec_zap.c:193: error: dereferencing pointer to incomplete type codec_zap.c:194: error: dereferencing pointer to incomplete type codec_zap.c:194: error: dereferencing pointer to incomplete type codec_zap.c:195: error: dereferencing pointer to incomplete type codec_zap.c:196: error: dereferencing pointer to incomplete type codec_zap.c:199: error: dereferencing pointer to incomplete type codec_zap.c:202: error: dereferencing pointer to incomplete type codec_zap.c:203: error: dereferencing pointer to incomplete type codec_zap.c:204: error: ?ZT_TCOP_TRANSCODE? undeclared (first use in this function) codec_zap.c:204: error: (Each undeclared identifier is reported only once codec_zap.c:204: error: for each function it appears in.) codec_zap.c:205: error: ?ZT_TRANSCODE_OP? undeclared (first use in this function) codec_zap.c: In function ?zap_destroy?: codec_zap.c:219: error: ?ZT_TCOP_RELEASE? undeclared (first use in this function) codec_zap.c:220: error: ?ZT_TRANSCODE_OP? undeclared (first use in this function) codec_zap.c:223: error: dereferencing pointer to incomplete type codec_zap.c: In function ?zap_new_alawtog723?: codec_zap.c:240: error: ?ZT_TCOP_ALLOCATE? undeclared (first use in this function) codec_zap.c:262: error: dereferencing pointer to incomplete type codec_zap.c:269: error: dereferencing pointer to incomplete type codec_zap.c:269: error: ?ZT_TRANSCODE_MAGIC? undeclared (first use in this function) codec_zap.c:270: error: dereferencing pointer to incomplete type codec_zap.c:271: error: dereferencing pointer to incomplete type codec_zap.c:277: error: dereferencing pointer to incomplete type codec_zap.c:278: error: dereferencing pointer to incomplete type codec_zap.c:279: error: ?ZT_TRANSCODE_OP? undeclared (first use in this function) codec_zap.c:281: error: dereferencing pointer to incomplete type codec_zap.c: In function ?zap_new_ulawtog723?: codec_zap.c:297: error: ?ZT_TCOP_ALLOCATE? undeclared (first use in this function) codec_zap.c:319: error: dereferencing pointer to incomplete type codec_zap.c:326: error: dereferencing pointer to incomplete type codec_zap.c:326: error: ?ZT_TRANSCODE_MAGIC? undeclared (first use in this function) codec_zap.c:327: error: dereferencing pointer to incomplete type codec_zap.c
Re: [asterisk-users] got-name
Bill Michaelson wrote: Is it just me, or is the AGI interface at cnam.got-name.com failing for others? Anyone know how to contact them without sending postal mail or telegram? I don't know how to contact them, but I am having the same problem. ___ --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] CallerID number not being displayed on SIP phones
I'm having trouble with Cisco 7960 and Linksys SPA-942 SIP phones not displaying the Caller-ID number. The Caller-ID name is displayed, but not the number. Instead, the phones always display the value that's set in the fromuser= parameter in sip.conf. If fromuser= is not set, then the literal asterisk is displayed in the calling number field on the telephone sets. Can I dynamically set the fromuser= value to the CallerID number in extensions.conf? How can I solve this problem? Asterisk v1.2.9 Cisco 7960 firmware P0S-3-07-4-00. ___ --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] Re: Rewriting caller ID from database?
Steven wrote: There are two I can think of. Hoodahek and asterdex (or asteridex) We used hoodahek at first, but now use asterdex(sp?) It has a web interface to enter the new names into. We use it to fixup, corp. cell phones and used to use it for our leagcy PBX extensions. I use some custom scripts to do database lookups and rewrite CallerID information. Everything works fine with regard to the CID name, however my Cisco 7960 and Linksys SPA-942 phones do not display the calling number. Instead, they display the called number. This makes the phone's call return feature not work. The calling number and name are both properly displayed on all of the softphone clients that I've tried. Here's the format I'm using to set the CallerID. SET CALLERID JONES DARYL A6508701826 Can anyone help? ___ --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] Need help with two-stage ringing macro
I've been using the following macro to ring SIP and IAX devices for a
few seconds, and then add on a cell phone if there is no answer on the
SIP or IAX device. Periodic problems began a few versions ago and now
the problem happens every time with 1.2.9 and 1.2.9.1.
The problem is that when a call from the PRI falls through to voicemail,
the call is dropped before the voicemail greeting is heard. Debug shows
that voicemail is starting and that Asterisk is dropping the call on the
PRI. Calls made from SIP or IAX devices work fine.
[macro-followme]
;
; modified standard extension macro for two-stage ringing.
;
; It will call the destinations in ${ARG4} for ${ARG2} seconds, and
; if that fails, the destinations in ${ARG5} for ${ARG3} seconds. If
; that also fails, it will send the call to voice mail for extension
; ${ARG1}.
;
; Note: if you want it to ring phone1 first, then phone1 AND phone2
; next, you have to list phone1 in both lists. Otherwise it will
; stop ringing on phone1.
;
; ${ARG1} - voice mail context
; ${ARG2} - Extension
; ${ARG3} - Time to ring stage 1
; ${ARG4} - Time to ring state 1 + 2
; ${ARG5} - Device(s) to ring stage 1
; ${ARG6} - Device(s) to ring stage 2
;
exten = s,1,SetCallerID(${CALLERIDNUM:-10:10}) ; Send only the last 10
digits
exten = s,2,NoOp(CallerID After:${CALLERIDNUM})
exten = s,3,SetAccount(${ARG2})
exten = s,4,Dial(${ARG5},${ARG3},rt) ; Ring the primary group
exten = s,5,Dial(${ARG5}${ARG6},${ARG4},rt) ; Add in the secondary group
exten = s,6,Voicemail([EMAIL PROTECTED]) ; send to vm as unavail
exten = s,7,Hangup
exten = s,106,Voicemail([EMAIL PROTECTED]) ; send to vm w/ busy
announce
exten = s,107,Hangup
___
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Re: [Asterisk-Users] Re: Asterisk and 7960s
The @ip-address is actually a documented cisco fix to another problem. I'd have to look it up, cause I don't remember exactly what it was, but it's been on the list somewhere, and I think EVERYONE that's used 8.2 has the same problem with the firmware. I would suggest using 7.4 or 7.5. I've been following this thread and it's clear is mud... Would someone care to summarize? Is it possible to automatically display the caller's number (true ANI in my case), caller name and caller address on a 7960 that's running 8.2? We currently rewrite CID Number and CID Name with a PHP script that does database lookups, however we can't get anything more than the name to display on the 7960. If this is possible, we sure would appreciate a summary of how you did it. ___ --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] * 1.0.8: no more reacting to callerid?
It's not just you. Same thing happens here. I went back to 1.0.7. Stefan Gofferje wrote: Hi folks, I used to have some constructions like exten = number/callerid,1,Goto(somewhere) After updating to 1.0.8 those does not work any more. Any hints? Regards, Stefan ___ 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
Re: [Asterisk-Users] DTMF stops working w/ Voicemail
Brent Franks wrote: I have some reports from users that occasionally DTMF will stop working in voicemail and they will have to exit the system to get it to work again. The useragents are Polycom IP 500's and I am using dtmfmode=rfc2833, with Ulaw codec. This is all on an internal switched 100mb lan. Has anyone else seen anything like this? Confirmed... Happens intermittently with Cisco 7960 phones for the past two weeks. I haven't been able to identify what causes it to occur. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Intermittent cidname lookups
I'm having a problem with intermittent lookup of Caller ID Name info using LookupCIDName. The same problem occurs when doing: asterisk -rx database show cidname No data is returned on every fourth or fifth query. No errors are being logged. I'm currently running CVS-HEAD-07/07/04-17:04:31 and first noticed the problem a few weeks ago. Is anyone seeing a similar problem? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FINALLY! a good book about Asterisk.
I've been reading drafts of this book for at least nine months and can assure you that its content is very different than what is available on the Wiki. The book is an excellent introduction to VOIP in general, and offers sufficient information for the novice to configure a basic Asterisk system. Advanced Asterisk users will still need the Wiki and mailing list archives. Perhaps a future edition of the book might cover more advanced topics, but the first edition is intended for beginners. Perhaps the most important thing that this book will accomplsih is to increase general awareness about Asterisk being a very reliable, full-featured PBX. Harold Workman wrote: what does that have to do with an overpriced book? and i agree with Joe. With this book sourcing most of the documentation directly from wiki, why pay for something thats free? Id rather donate $49 to keeping wiki free to the enviroment. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WAMi - Windows Asterisk Manager
The default password is Admin. Adam Goryachev wrote: On Tue, 2004-04-06 at 14:36, Christian Hoffmeyer wrote: Thank you for all of the beta testing. New and improved graphics in this release along with drag and drop transfers and hold for all technologies. There's a screenshot on the link below. Also improved documentation so read the included README. There's also a sample xml configuration included. http://www.voip-info.org/tiki-index.php?page=Asterisk+WAMI I can't seem to find where I am supposed to create the config file, nor do I know what the default admin password is.. Any suggestions? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One voicemail - multiple boxes?
I contracted with Digium for this enhancement and am waiting for it to be completed. Tilghman Lesher wrote: On 2004 Apr 02, at 12:04, Brian Capouch wrote: I don't want to re-invent the wheel if someone has already hacked a way to do this. One of my customers has a number of stores, and he wants to leave one voicemail that would be delivered to all the managers at once. Each has a voicemail account on his server. I have googled around and looked on the WIKI. Maybe I'm missing it? There's a request to do this on the bugtracker, but no implementations yet, AFAIK. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk agi interface leaves zombie processes?
This is a known problem. I have the same situation with RH9 as you do. I don't know if the problem has been added to the new bug tracking system. We should check. My workaround is to run the AGI scripts on a RH7 box and forward calls using IAX. Scott Stingel wrote: Hi- Asterisk (CVS 7/30/03) seems to leave my AGI processes (written in Perl) as zombies, even though they exit normally with exit(0). I am running Red Hat 9. I tried the same AGI etc with an older CVS (7/1/03) and this does not happen. I think a zombie process is a process that doesn't release some resource (shared memory, etc), so I'm worried about this since I call my AGI once per incoming call. Has anyone else experienced this? Thanks Scott Scott M. Stingel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail enhancements
Brad's recent list of enhancements look good, but I haven't looked at the code yet. If the code looks good, I hope it will be committed to the project CVS. Here's a partial list of enhancements that I would like to see in Comedian Mail. I am probably interested in helping to fund the enhancement of Asterisk voicemail. Is anyone else interested? -Address message to multiple recipients -Forward message to multiple recipients -Recipient groups maintained by a user-accessible web page -Scheduled future deliver -Say digits of calling number when requested -Reminder paging notification until message is retrieved ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Robbed bit signalling debugging
I'm trying to debug a problem with robbed bit signalling on a T1 coming into an Asterisk box on a T100P card. Specifically, I need to look at the signalling timing. Is there a way to turn on this kind of debugging in Asterisk, similar to what 'pri debug' does? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using asterisk for a 911 call center....
911 trunks are usually delivered to public-safety answering points (PSAP) on analog reverse-battery facilities. (The PSAP provides battery toward the CO). ANI is provided using MF tones. The PSAP equipment must take the ANI and use it to submit a database query to lookup the caller's address (ALI - automatic location identification) to an external database that is usually operated by the ILEC or its contractor. The ANI and ALI information must be presented to the 911 call taker immediately upon answering the call. There is definitely a place for Asterisk in the public-safety telecomm field, but a lot of work would be needed for it to handle calls in a PSAP. If you're interested in learning more about PSAP equipment, take a look at the web pages for Plant Equipment, Inc, Positron, Inc. and Zetron, Inc. On Mon, 21 Jul 2003, Gene Kochanowsky wrote: Has anyone had any experience using asterisk for a 911 call center? Does anyone know of any reason why it would not be suitable? As far as I know all 911 call routing takes place at the CO switch so a regular T1 line should work fine. I understand that there is support for ACD in asterisk and that is should be possible to implement screen pop (CTI). Any comments? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] EM DID config question
I am trying to make an in/out trunk group comprised of 4 DS0's using EM Wink signalling. The first four channels of a DS1 on a T100P are being used for the group. Outbound calls work fine, but inbound calls fail. The other 20 DS0 channels are used for a PRI. Does the configuration shown below look okay? I've tried setting 'immediate = yes' without success, but it doesn't seem to make any difference.. It seems like Asterisk never gets any digits from the upstream switch. I don't think the upstream switch gets a wink from Asterisk, but I am not sure. Here's what the console log shows. -- Starting simple switch on 'Zap/1-1' File chan_zap.c, Line 3772 (ss_thread): getdtmf on channel 1: Operation now in progress == Unknown extension 's' in context 'default' requested -- Playing 'ss-noservice' -- Hungup 'Zap/1-1' Incidentally, inbound calls on the PRI are immediately disconnected when inbound caller-id info is present. The EM trunk group is an attempt to workaround this problem. = zapata.conf [channels] group = 1 context = default switchtype = national signalling = pri_cpe callerid = asreceived amaflags = billing pri_dialplan = national echocancelwhenbridged = yes echocancel = 128 channel = 5-23 group = 2 signalling = em_w context = default immediate = no amaflags = billing echocancelwhenbridged = yes channel = 1-4 = zaptel.conf span=1,0,0,esf,b8zs bchan=5-23 dchan=24 loadzone=us defaultzone=us em=1-4 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] EM DID config question
Thanks for the info. My 'telco' is an Adtran Atlas that I have management control of. I broke out the 4 DS0's into a separate trunk group for testing. I don't see a way to configure the Atlas to not send caller-id info on outbound PRI channels, but will look further. Eventually, I need caller-id info to accomplsih my goal. I'll try adding D channel def in zapata.conf. I don't expect to get caller-id info on a B channel. On Sat, 5 Jul 2003, Steven Critchfield wrote: First off, caller ID should be in the q.931 packets and not on the B channels of a PRI. So if the fsk spill is causing problems, go back to full PRI and turn off callerid from your telco. One thing I noticed below is you don't have your D channel defined in zapata.conf. I think for em you need immediate= no. It appears that from your message below that you are picking up the line, and you are not getting DTMF so it is trying to go to a s extension that doesn't exist. It seems odd that the telco would split a PRI to give EM on the same T1. If they aren't doing that, it would explain the lack of DTMF, but then I don't think you would get ring events. Ring events for a PRI are in the D channel where EM are in the robbed bit. On Sat, 2003-07-05 at 13:57, Daryl Jones wrote: I am trying to make an in/out trunk group comprised of 4 DS0's using EM Wink signalling. The first four channels of a DS1 on a T100P are being used for the group. Outbound calls work fine, but inbound calls fail. The other 20 DS0 channels are used for a PRI. Does the configuration shown below look okay? I've tried setting 'immediate = yes' without success, but it doesn't seem to make any difference.. It seems like Asterisk never gets any digits from the upstream switch. I don't think the upstream switch gets a wink from Asterisk, but I am not sure. Here's what the console log shows. -- Starting simple switch on 'Zap/1-1' File chan_zap.c, Line 3772 (ss_thread): getdtmf on channel 1: Operation now in progress == Unknown extension 's' in context 'default' requested -- Playing 'ss-noservice' -- Hungup 'Zap/1-1' Incidentally, inbound calls on the PRI are immediately disconnected when inbound caller-id info is present. The EM trunk group is an attempt to workaround this problem. = zapata.conf [channels] group = 1 context = default switchtype = national signalling = pri_cpe callerid = asreceived amaflags = billing pri_dialplan = national echocancelwhenbridged = yes echocancel = 128 channel = 5-23 group = 2 signalling = em_w context = default immediate = no amaflags = billing echocancelwhenbridged = yes channel = 1-4 = zaptel.conf span=1,0,0,esf,b8zs bchan=5-23 dchan=24 loadzone=us defaultzone=us em=1-4 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA-186 de-register
Yes, but I have been able to mitigate it by setting the following parameters. I have the problem with ATA's that are behind firewalls and not, but mostly with the ones that are behind firewalls. CfgInterval:1800 SIPRegInterval:100 On Thu, 3 Jul 2003, Kim C. Callis wrote: Is it just me or do others have a problem with the ATA-186 de-registering? Every couple of hours, if I don't make use of the ATA connected line, I find that I have to unplug and let the ATA reboot. After that it is good to go for awhile, but eventually I have to repeat the process. My ATA sits behind a NATd firewall, any ideas what might cause the de-registration? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What user-id should Asterisk run under
Should Asterisk run under it's own user id, or the web server user id, or root, or what? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with zombies left after calls to Festival
I started using Festival for the first time today and am having a problem with zombies left behind after every time that it speaks. I'm using Festival 1.4.3 with today's CVS of Asterisk. Everything seems to work. The only obvious problem is that a defunct process is left behind every call to Festival. Is this a known problem? Does anyone know how I can fix this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need help with inbound/outbound PRI calls
I'm running a pretty successful Asterisk system and recently moved our PRI to a T100P board. The PRI was previously connected to a Cisco 2600 that was serving as a voice gateway. We are having a frequent problem with inbound and outbound calls being disconnected shortly after they are answered since moving the PRI directly to the Asterisk box. Most calls work fine, but approx 3 out 10 are prematurely disconnected. Debug info from an outbound call is included below. Note that the disconnect cause is 69, which I believe means Requested facility not implemented. Am I interpreting the debug info correctly? What would cause this? We had a similar problem with the Cisco gateway, but worked around it by configuring the Cisco to force the numbering plan for outgoing calls from 'unknown' to 'national'. Is there a way to do this in Asterisk? -- debugging info from dropped outbound call --- Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Unknown (1) '6505901801' ] Called Number (len=14) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '15592982000' ] -- Called g1/15592982000 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 32802/0x8022) (Terminator) Message type: CALL PROCEEDING (2) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (Channel Identification) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32802/0x8022) (Terminator) Message type: ALERTING (1) Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Processing IE 30 (Progress Indicator) -- Zap/1-1 is ringing -- Registered SIP '99051' at 209.234.100.2 port 1175 expires 120 Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 32802/0x8022) (Terminator) Message type: CONNECT (7) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 34/0x22) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/1-1 answered SIP/99050-a3fe Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 34/0x22) (Originator) Message type: DISCONNECT (69) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' == Spawn extension (trusted, 15592982000, 1) exited non-zero on 'SIP/99050-a3fe' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 32802/0x8022) (Terminator) Message type: RELEASE (77) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 34/0x22) (Originator) Message type: RELEASE COMPLETE (90) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Directory Application question
This doesn't work for me. Voicemail says the extension number but does not play the user's name. (Asterisk CVS-04/30/03-22:57:49) On Tue, 17 Jun 2003, Benjamin Miller wrote: When in voicemail they need to go into the record name section and record their name. Then it will play their name. -Original Message- From: Derek Beaumont [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 17, 2003 3:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Directory Application question I'm wondering if I can do the following: Caller activates the Directory application Caller enters the first 3 digits of a person's last name = Normally here, Asterisk will say the extension number of a person found. Is there a way to get Asterisk to say the name as well? (perhaps using the same sound file that is used for their name in the voicemail application) Can this be done? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Reminder paging for voicemail (?)
Is there a way to configure voicemail to do reminder paging? I would like to configure some voicemail boxes to send an e-mail message to a pager every 10 minutes until the message is retrieved. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip channel driver causes asterisk to crash when talking to quintum A800
I experienced the exact same symptoms but didn't have the confidence to post my experience to this list because of my lack of experience with Asterisk. I restored the June 1 version from CVS and the problem went away. There's definitely a problem in code since June 1. On Sat, 7 Jun 2003, John Todd wrote: - After that point, all other SIP calls from any other device fail, and looking at tethereal I see that there are no replies to new SIP REGISTER requests, either. I can type stop now or stop gracefully and the system will not stop. I have to manually killall to get asterisk to die. - I backed out to a version from June 3 21:18 and all dial modes work correctly with exactly the same /etc/asterisk/* files, so it is a change in Asterisk and not in the phones. *CLI show version Asterisk CVS-06/07/03-01:40:15 built by [EMAIL PROTECTED] on a i686 running Linux *CLI *CLI sip debug SIP Debugging Enabled Sip read: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.1.25:5060 From: sip:[EMAIL PROTECTED];user=phone;tag=2961659159 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp User-Agent: Cisco ATA 186 v2.16 ata18x (030401a) Expires: 300 Content-Length: 243 Content-Type: application/sdp v=0 o=2204 23257 23257 IN IP4 10.0.1.25 s=ATA186 Call c=IN IP4 10.0.1.25 t=0 0 m=audio 16386 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 11 headers, 11 lines Using latest request as basis request Sending to 10.0.1.25 : 5060 (non-NAT) Capabilities: us - 14, them - 268, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 *CLI *CLI *CLI show channels Channel (ContextExtensionPri ) State Appl. Data 0 active channel(s) *CLI *CLI *CLI sip show channels Peer UsernameCall ID Seq (Tx/Rx) Lag Jitter Format 10.0.1.25(None) 1852710522@ 00101/2 0ms ms 0 1 active SIP channel(s) *CLI Configuration for ATA-186 line 1: [2204] type=friend username=2204 secret=somepassword mailbox=2203 host=dynamic context=intern canreinvite=no dtmfmode=rfc2833 nat=1 For reference, here is the SIP debug for a functional call from a 7960 on the same version of Asterisk code (2203 = 7960, 2204 = ATA-186 line 1) *CLI Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.1.15:5060 From: 2203 sip:[EMAIL PROTECTED];tag=0002b9eb0ef400c3289c4132-36211630 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Date: Sat, 07 Jun 2003 19:19:33 GMT CSeq: 101 INVITE User-Agent: CSCO/4 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Content-Type: application/sdp Content-Length: 241 Accept: application/sdp Remote-Party-ID: 2203 sip:[EMAIL PROTECTED];party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 14891 19200 IN IP4 10.0.1.15 s=SIP Call c=IN IP4 10.0.1.15 t=0 0 m=audio 23764 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 14 headers, 11 lines Using latest request as basis request Sending to 10.0.1.15 : 5060 (non-NAT) Capabilities: us - 14, them - 268, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
