[asterisk-users] Newcomer Question
Hallo Group! My Name is Guenther Sohler and I registred to this group, because I think asterisk could be interesting for me. I have got a small server at home running linux. It does NAT and a Firewall. There is an intranet with my home PC and a hardware SIP phone. This SIP phone registers at mujtelefon.cz Now I got another account at sipgate.at My idea is following: I want to be reachable at both providers(numbers) at the same time. And If I call someone, calls to austria shall use sipgate, whereas calls to czech shall use mujtelefon. So far, i read through the tutorial at digium.com and got the impression, that asterisk might be able to do this and makes phoning very conveniant. YEsterday evening i managed to get a compiled version of asterisk running on my server. I just font yet have a complete idea, what is to be changed * get asterisk running on my server as phone central(registrar) * which firewall settings ? before/after nat * my hardware phone registers with asterisk at my server * which files to i have to change ? dialplan, sip.conf? How do I achieve this ? Thanx in advance! -- Pt! Schon vom neuen GMX MultiMessenger gehört? Der kanns mit allen: http://www.gmx.net/de/go/multimessenger ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newcomer Question
hi, and first off all ... welcome! now it would be handy if you provide us with the name of your phone for ex 'a linksys spa942' or somthing kr, Jan de Coster Guenther Sohler wrote: Hallo Group! My Name is Guenther Sohler and I registred to this group, because I think asterisk could be interesting for me. I have got a small server at home running linux. It does NAT and a Firewall. There is an intranet with my home PC and a hardware SIP phone. This SIP phone registers at mujtelefon.cz Now I got another account at sipgate.at My idea is following: I want to be reachable at both providers(numbers) at the same time. And If I call someone, calls to austria shall use sipgate, whereas calls to czech shall use mujtelefon. So far, i read through the tutorial at digium.com and got the impression, that asterisk might be able to do this and makes phoning very conveniant. YEsterday evening i managed to get a compiled version of asterisk running on my server. I just font yet have a complete idea, what is to be changed * get asterisk running on my server as phone central(registrar) * which firewall settings ? before/after nat * my hardware phone registers with asterisk at my server * which files to i have to change ? dialplan, sip.conf? How do I achieve this ? Thanx in advance! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newcomer Question
My Phone identifies as USer-Agent: ALL7950 02.09.23 I suppose its AllNet 7950 Hope this helps :) Original-Nachricht Datum: Thu, 20 Sep 2007 08:36:59 +0200 Von: Jan De Coster [EMAIL PROTECTED] An: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Betreff: Re: [asterisk-users] Newcomer Question hi, and first off all ... welcome! now it would be handy if you provide us with the name of your phone for ex 'a linksys spa942' or somthing kr, Jan de Coster Guenther Sohler wrote: Hallo Group! My Name is Guenther Sohler and I registred to this group, because I think asterisk could be interesting for me. I have got a small server at home running linux. It does NAT and a Firewall. There is an intranet with my home PC and a hardware SIP phone. This SIP phone registers at mujtelefon.cz Now I got another account at sipgate.at My idea is following: I want to be reachable at both providers(numbers) at the same time. And If I call someone, calls to austria shall use sipgate, whereas calls to czech shall use mujtelefon. So far, i read through the tutorial at digium.com and got the impression, that asterisk might be able to do this and makes phoning very conveniant. YEsterday evening i managed to get a compiled version of asterisk running on my server. I just font yet have a complete idea, what is to be changed * get asterisk running on my server as phone central(registrar) * which firewall settings ? before/after nat * my hardware phone registers with asterisk at my server * which files to i have to change ? dialplan, sip.conf? How do I achieve this ? Thanx in advance! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Der GMX SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! Ideal für Modem und ISDN: http://www.gmx.net/de/go/smartsurfer ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk crash and core dump: format_mp3.so
Hi, My Asterisk server process irregularly segfaults, ie. it usually works fine (is stable) when there's low traffic but repeatedly crashes during morning hours when there are more calls. I gdb'ed the core dump files and found that the culprit may be format_mp3. So I disabled MOH today and will see if that's the cause. I know that mp3 files are known to cause * crashes but what I don't understand is why it doesn't *always* crash (ie. why doesn't it crash even when there's low traffic? I mean, if the offending code is in the mp3 format then it should *always* crash, right?). I'm pasting the bt below to see if someone has any suggestions on this ML. I'd rather wait a little before bothering the devel list... Core was generated by `/usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg'. Program terminated with signal 11, Segmentation fault. #0 0xb7b41954 in ?? () from /usr/lib/asterisk/modules/format_mp3.so (gdb) bt #0 0xb7b41954 in ?? () from /usr/lib/asterisk/modules/format_mp3.so #1 0xb7ea6ff4 in ?? () from /lib/libc.so.6 #2 0xb7ea8120 in ?? () from /lib/libc.so.6 #3 0x001a in ?? () #4 0x in ?? () (gdb) bt full #0 0xb7b41954 in ?? () from /usr/lib/asterisk/modules/format_mp3.so No symbol table info available. #1 0xb7ea6ff4 in ?? () from /lib/libc.so.6 No symbol table info available. #2 0xb7ea8120 in ?? () from /lib/libc.so.6 No symbol table info available. #3 0x001a in ?? () No symbol table info available. #4 0x in ?? () No symbol table info available. (gdb) (gdb) thread apply all bt ... not posting because too long here ... Thanks, Vieri Shape Yahoo! in your own image. Join our Network Research Panel today! http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] loop detected
I have an asterisk 1.4, that was working properly, but from last week, without any changing in the config of asterisk, all of calls,fall in loop detected error. there is two strange actions: 1-the first call after restarting the asterisk, is done successfully . 2-no packet , was sent to the destination, during this error. plz help me best Mani Shape Yahoo! in your own image. Join our Network Research Panel today! http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk crash and core dump: format_mp3.so
On Thursday 20 September 2007 11:34:44 Vieri wrote: My Asterisk server process irregularly segfaults, ie. it usually works fine (is stable) when there's low traffic but repeatedly crashes during morning hours when there are more calls. I gdb'ed the core dump files and found that the culprit may be format_mp3. So I disabled MOH today and will see if that's the cause. I know that mp3 files are known to cause * crashes but what I don't understand is why it doesn't *always* crash (ie. why doesn't it crash even when there's low traffic? I mean, if the offending code is in the mp3 format then it should *always* crash, right?). We also experienced this problem on 1.2, but i'm not sure that this is registered in bug database. You should check bugs.digium.com and if it's still valid for 1.4, you should post your backtraces there. As solution - we refused from using format_mp3 at all - actually it has almost no benefits. If your MOH is in MP3s - you will get them decoded (and translated to necessary codec) on-the-fly for every call, so more performance. You can convert all your MOH to native channell formats of asterisk, and put all those files (one for each format/MOH combination) in MOH directory - asterisk will pick up one with less translation. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk crash and core dump: format_mp3.so
--- Atis Lezdins [EMAIL PROTECTED] wrote: We also experienced this problem on 1.2, but i'm not sure that this is registered in bug database. You should check bugs.digium.com and if it's still valid for 1.4, you should post your backtraces there. Actually, I'm using 1.2.21.1 so since 1.2 will only receive security fixes I don't think I'll post a bug report. As solution - convert all your MOH to native channell formats of asterisk, and put all those files (one for each format/MOH combination) in MOH directory - asterisk will pick up one with less translation. Thanks Atis. Will try that out. Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newcomer Question
Am Donnerstag, den 20.09.2007, 08:30 +0200 schrieb Guenther Sohler: Hallo Group! My Name is Guenther Sohler and I registred to this group, because I think asterisk could be interesting for me. Hi Guenther, this place probably is the right one. Welcome! I have got a small server at home running linux. It does NAT and a Firewall. There is an intranet with my home PC and a hardware SIP phone. This SIP phone registers at mujtelefon.cz Now I got another account at sipgate.at My idea is following: I want to be reachable at both providers(numbers) at the same time. And If I call someone, calls to austria shall use sipgate, whereas calls to czech shall use mujtelefon. This is possible, and it does not require too difficult steps. First question though is wether your server has an external IP (e.g. does the internet routing) or there is a router in between (you wrote the server does NAT, but I already saw double- and even triple-NAT configurations - I have to mention that). Both will work, but _not_ having NAT in between might be one trouble source less - so if you run Asterisk on a machine with a globally valid and routable IP, you are better off. Your firewall should accept incoming TCP on port 5060 and incoming UDP on all the ports RTP uses (like 1 to 2) - I rarely bother firewalling incoming UDP packets on high ports, but you should check that. If your phone works behind the router, the UDP requirement probably is already sorted. Basically, you will have to edit a few configuration files. I will give some examples based on one of my asterisk configs, but you really should read about those files and check wether everything is OK - I will try to adapt to your situation, but do not blame me if I mistype or just mis-think something. In sip.conf, you will need to list the providers and the phones you are going to use. I assume you will have your allnet and perhaps a few softphones - you will probably want more than one phone some day ;-) 8 sip.conf (with example data indicated) [general] bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm musicclass=default language=en ; well, no idea if there are czech audio files readily available. ; I personally use language=de, of course. dtmfmode=rfc2833 sipdebug=no register = 1234567:[EMAIL PROTECTED]:5060/004311234567 ; put your sip id (1234567), password (4321) and your ; phone number (004311234567) here register = 123321321:[EMAIL PROTECTED]:5060/12 [sipgateat] host=sipgate.at secret=4321 username=1234567 fromuser=1234567 fromdomain=sipgate.at srvlookup=yes context=sipgateat-in canreinvite=no nat=no ; perhaps this needs to be set to yes ; insecure=very ; perhaps this needs to be activated - try it. type=friend qualify=yes [otherprovider] host=otherprovider.example.org secret=abcd username=123321321 fromuser=123321321 fromdomain=otherprovider.example.org srvlookup=yes context=otherprovider-in canreinvite=no nat=no type=friend qualify=yes ; stanza for SIP clients [sip01] mailbox=01 callerid=11 type=friend username=sip01 secret=LaBananaLoca ; replace with the secret for your telephone, username should ; always be the same as the [stanza] name to avoid trouble context=sipclient host=dynamic nat=yes [sip02] mailbox=01 callerid=12 type=friend username=sip01 secret=AyayayDiosMio context=sipclient host=dynamic nat=yes 8 so much for the sip.conf. This allows for two accounts with providers, and two SIP phones (wether hard- or softphone does not matter, of course :-) You will also need to setup an extensions.conf, somehow like this 8 extensions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no ;; all of those have been like this in my conf for ages, I do not ;; even know what exactly those are good for. ; context where sipclient outgoing calls are handled [sipclient] ; let 11 and 12 be internal numbers exten = 11,1,Dial(SIP/sip01,60) exten = 11,2,Hangup() exten = 12,1,Dial(SIP/sip02,60) exten = 12,2,Hangup() ; Outward calls. If a country prefix is present _and_ it is Austria, ; use sipgate.at exten = _0043.,1,Dial(SIP/[EMAIL PROTECTED],60) exten = _0043.,2,Hangup() ; Outward calls with country prefix for Czech Republic go through ; your other provider exten = _00420.,1,Dial(SIP/[EMAIL PROTECTED],60) exten = _00420.,2,Hangup() ; All other non-international calls go through otherprovider - ; three digit minimum here, shorter numbers treated as internal exten = _0[1-9].,1,Dial(SIP/[EMAIL PROTECTED],60) exten = _0[1-9].,2,Hangup() exten = _[1-9][0-9].,1,Dial(SIP/[EMAIL PROTECTED],60) exten = _[1-9][0-9].,2,Hangup ; add stuff for voicemail call-in here ; context for incoming calls through sipgate [sipgateat-in] exten = 004311234567,1,Dial(SIP/sip01SIP/sip02,60) exten = 004311234567,2,Hangup() [otherprovider-in] exten = 12,1,Dial(SIP/sip01SIP/sip02,60) exten = 12,2,Hangup() 8 This should get you started. This is a very rough example, and I
[asterisk-users] Horrible problem - calls losing sound
We're having a horrid problem with our asterisk setup. Sometimes calls just go dead - we can't hear what the other end is saying. (I think they can't hear us either). The call doesn't hang up until one of the callers gets bored. Internaly we use Thomson ST2030 SIP phones. Externaly we have 3 ISDN BRI lines (6 channels total), connected to an Eicon Diver server card (4BRI). We're using Asterisk 1.4.11 (1.4.11-BRIstuffed-0.4.0-test4) on a Debian system, with chan-capi 1.0.1. Any idea what could be going wrong, where to look c? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G.722: ast_channel_make_compatible failure
Hi all, I have an interesting problem with Asterisk 1.4.11 - 3 SIP phones: [phone1] allow=g722 allow=alaw [phone2] allow=alaw allow=g722 [phone3] allow=alaw Now, when I try to call: 1. phone1 calling phone2, I got through, using G.722 codec 2. phone2 calling phone1, I get through, using Alaw 3. phone3 calling phone1 or phone2, OK using Alaw But: 4. phone1 calling phone3 fails: [Sep 20 10:14:32] WARNING[30706]: chan_sip.c:2963 sip_call: No audio format found to offer. Cancelling call to phone3 Any ideas what could be wrong? Many thanks for any suggestion Ondrej The information contained in this e-mail and in any attachments is confidential and is designated solely for the attention of the intended recipient(s). If you are not an intended recipient, you must not use, disclose, copy, distribute or retain this e-mail or any part thereof. If you have received this e-mail in error, please notify the sender by return e-mail and delete all copies of this e-mail from your computer system(s). Please direct any additional queries to: [EMAIL PROTECTED] Thank You. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux-HA and Asterisk
Router modell number ? On a linksys or netgear on incoming calls the wrong phones start ringing (unless the router is sip aware) - Original Message - From: Jerry Jones [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 12, 2007 3:18 PM Subject: Re: [asterisk-users] Linux-HA and Asterisk How about 20+ on a Qwest DSL modem hitting our server? Works great. On Sep 12, 2007, at 7:23 AM, Dovid B wrote: Eric, Try 5 polycoms behind the same NAT router. Let me know when you grab a drink ;) - Original Message - From: Eric ManxPower Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 12, 2007 2:43 PM Subject: Re: [asterisk-users] Linux-HA and Asterisk Polycoms work just fine behind NAT. Mike Clark wrote: Chris Mason (Lists) wrote: Mike Clark wrote: Yes, the Asterisk boxes were on private addresses. The Polycoms are also behind a NAT. Yes, I tried using externip in sip.conf and this allowed registration, and calls to be placed, but no audio. Unfortunately, Polycom does not support STUN. Your problem is not Linux-HA, it looks like that is fully functional. Your problem is the same one many people come across. You can't put Polycom phones behind NAT, it won't work. If you have to have the phones behind NAT, which I advise against, use Linksys which probably work better, and use a SIP aware NAT device. Better still, put the phones on the same network as the Asterisk PBX and say goodbye to your problems. Thanks Chris. Unfortunately, these solutions aren't an option. I guess I was hoping someone had found the silver bullet or some undocumented Asterisk feature that solved the issue. Back to the drawing board. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http:// www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Horrible problem - calls losing sound
Hi John! I have the same problem, the system contains 1 port Billion ISDN BRI card, and 1 sip trunk. This is a trixbox with Asterisk 1.2.22-BRIstuffed-0.3.0-PRE-1y-i The ISDN call is forwarded to a ring-group. The 6 sip phones are welltech lp399 series. If incoming the call get wrong, we can not hear the other side, but they hear us. In my case the rtp debug shows there are no incoming rtp packets from asterisk to SIP phone. If somebody experienced this problem, please help US! Thanks! 2007/9/20, John Hughes [EMAIL PROTECTED]: We're having a horrid problem with our asterisk setup. Sometimes calls just go dead - we can't hear what the other end is saying. (I think they can't hear us either). The call doesn't hang up until one of the callers gets bored. Internaly we use Thomson ST2030 SIP phones. Externaly we have 3 ISDN BRI lines (6 channels total), connected to an Eicon Diver server card (4BRI). We're using Asterisk 1.4.11 (1.4.11-BRIstuffed-0.4.0-test4) on a Debian system, with chan-capi 1.0.1. Any idea what could be going wrong, where to look c? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ Tóth Péter Tel.: +36703834578 _ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Horrible problem - calls losing sound
On Thu, Sep 20, 2007 at 01:24:52PM +0200, Péter Tóth wrote: Hi John! I have the same problem, the system contains 1 port Billion ISDN BRI card, and 1 sip trunk. This is a trixbox with Asterisk 1.2.22-BRIstuffed-0.3.0-PRE-1y-i The ISDN call is forwarded to a ring-group. The 6 sip phones are welltech lp399 series. If incoming the call get wrong, we can not hear the other side, but they hear us. In my case the rtp debug shows there are no incoming rtp packets from asterisk to SIP phone. If somebody experienced this problem, please help US! Just as you have rtp debug, you have bri debug . bri debug span 1 and hope for a friendly ISDN guru on the list... -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Building an RPM from Asterisk 1.4
On Thu, Sep 20, 2007 at 12:22:43AM +0200, Tzafrir Cohen wrote: Why is it looking for files that obviously don't exist? That spec uses quite a few discourged methods for rpm packages. There are a number of well-maintained RPM packages of Asterisk. Use one of them or modify one of them. Do you have any examples for these spec files? I found a repository for installing Asterisk on Centos, but it took a while before I discovered it. Ok, just checked the link its for RHEL, but as Centos is just recompiled this won't matter. Same situation for Ubuntu using the debian package format, but I have not found a repository so far and Ubuntu delivers just the old 1.2 release. :) Marcus ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems sending more than 2 SMS with asterisk / smsq
On 9/19/07, Christoph Adomeit [EMAIL PROTECTED] wrote: Especially I do not see how I could add a wait to the dialplan as somebody suggested because there seems no dialplan invoked when I send sms. Can you not invoke a shell script and put the sleep in there? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ghost calls from phones
Hi all, I have since days now a strange problem with two Thomson ST2030 phones (FMW 3.56) on Asterisk 1.4.11. They are both in a queue (only one phone per queue to get the MoH played ...) No i see often times in the CDR these: 33012 2007-09-20 14:16:52+02 s outgoing-staff SIP/test1-082ca1c8 6 0 NO ANSWER 3 1190290612.216 So calls to dst s and always different SIP channels. During that calls I Get back on the CLI a -- Got SIP response 486 Busy back from 172.10.4.170 -- SIP/reservation-b5c322c8 is busy so the phone seems to stuck in calls and is not reachable anymore Any idea? Kind Regards, Erik ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Horrible problem - calls losing sound
On Thu, 2007-09-20 at 12:49 +0200, John Hughes wrote: We're having a horrid problem with our asterisk setup. Sometimes calls just go dead - we can't hear what the other end is saying. (I think they can't hear us either). The call doesn't hang up until one of the callers gets bored. Internaly we use Thomson ST2030 SIP phones. Externaly we have 3 ISDN BRI lines (6 channels total), connected to an Eicon Diver server card (4BRI). We're using Asterisk 1.4.11 (1.4.11-BRIstuffed-0.4.0-test4) on a Debian system, with chan-capi 1.0.1. Any idea what could be going wrong, where to look c? I'm don't know what is causing your problem but afaik you don't need bristuff with your Eicon Diva Server card. I have several asterisk 1.2.24 boxes with 1x BRI and 4x BRI Eicon Diva Server cards and I only use asterisk + chan_capi. This setup is very stable and works great. Perhaps you could try rebuilding asterisk without the bristuff patch then rebuild and install chan_capi and see if the problem goes away? Regards, Patrick ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Horrible problem - calls losing sound
Tzafrir Cohen wrote: On Thu, Sep 20, 2007 at 01:24:52PM +0200, Péter Tóth wrote: Just as you have rtp debug, you have bri debug . bri debug span 1 and hope for a friendly ISDN guru on the list... Does nothing for me - my isdn is connected via chan-capi. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Building an RPM from Asterisk 1.4
On Thu, Sep 20, 2007 at 02:25:19PM +0200, Marcus Franke wrote: Same situation for Ubuntu using the debian package format, but I have not found a repository so far and Ubuntu delivers just the old 1.2 release. :) Ubuntu packages of Asterisk are slightly modified Debian ones. As for the Debian ones: http://packages.debian.org/asterisk As you can see there, the Unstable packages are currently at 1.4.11 . Not that bad. There are certain petty things that delay that package from getting into Testing right now. Also in that page you an find the sources of the package (tarball, dif and a small .dsc file). http://packages.debian.org/source/sid/asterisk will also give you link to the subversion where the sources (or rather: the diff) are maintained. And then there's a build server where new and untested packages are built. Including backports for various Debian and Ubuntu distributions. http://buildserver.net/ -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems sending more than 2 SMS with asterisk / smsq
Am Mittwoch, den 19.09.2007, 15:25 +0200 schrieb Christoph Adomeit: Hi there, I experience the same problem here with asterisk 1.2.24 on an E1 Line, only 2 of 3 sms are sent, this happens always and is reproducable. Did someone find out more about the problem ? Especially I do not see how I could add a wait to the dialplan as somebody suggested because there seems no dialplan invoked when I send sms. I use: smsq -d 017xxx -m TEST1 --motx-channel=Zap/g1/0193010 (Germen Telekom Message Center) How could I invoke smsq differently to use an own context of the dialplan ? I think I have been in error there. The wait occurs on incoming calls, like my Gigaset calling in to Asterisk, between answering the line: exten = _0193010.,1,Answer() exten = _0193010.,2,Wait(2) exten = _0193010.,3,SMS(blahfasel) Same for _incoming_ messages from the Telco SMSC. I do not immediately understand where I could have inserted a Wait() for outgoing SMS, especially as that SMS() seems to open the line itself. I would have to investigate, but do not have the equipment right here at the moment, sorry. BR Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Horrible problem - calls losing sound
Tzafrir Cohen wrote: On Thu, Sep 20, 2007 at 01:24:52PM +0200, Péter Tóth wrote: Hi John! I have the same problem, the system contains 1 port Billion ISDN BRI card, and 1 sip trunk. This is a trixbox with Asterisk 1.2.22-BRIstuffed-0.3.0-PRE-1y-i The ISDN call is forwarded to a ring-group. The 6 sip phones are welltech lp399 series. If incoming the call get wrong, we can not hear the other side, but they hear us. In my case the rtp debug shows there are no incoming rtp packets from asterisk to SIP phone. If somebody experienced this problem, please help US! Just as you have rtp debug, you have bri debug . bri debug span 1 and hope for a friendly ISDN guru on the list... Ah, I can get the same (excessive!) info by set verbose 8 capi debug ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Horrible problem - calls losing sound
Patrick wrote: On Thu, 2007-09-20 at 12:49 +0200, John Hughes wrote: We're having a horrid problem with our asterisk setup. Sometimes calls just go dead - we can't hear what the other end is saying. (I think they can't hear us either). The call doesn't hang up until one of the callers gets bored. Internaly we use Thomson ST2030 SIP phones. Externaly we have 3 ISDN BRI lines (6 channels total), connected to an Eicon Diver server card (4BRI). We're using Asterisk 1.4.11 (1.4.11-BRIstuffed-0.4.0-test4) on a Debian system, with chan-capi 1.0.1. Any idea what could be going wrong, where to look c? I'm don't know what is causing your problem but afaik you don't need bristuff with your Eicon Diva Server card. I have several asterisk 1.2.24 boxes with 1x BRI and 4x BRI Eicon Diva Server cards and I only use asterisk + chan_capi. This setup is very stable and works great. Perhaps you could try rebuilding asterisk without the bristuff patch then rebuild and install chan_capi and see if the problem goes away? Yeah, I was being lazy, just using the asterisk from the Debian repositories. Guess I'll try with my own build. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints / State change on outgoing calls
On 9/19/07, Alex Epshteyn [EMAIL PROTECTED] wrote: Also, Asterisk restart results in all the watchers being lost. Is there a way to force the phone to subscribe to notifications after restart (short of rebooting it) and is it phone specific? Usually resubscribe-interval for extensions is client controlled, much like SIP re-register interval. Just make sure it's in between the min and max registration times as displayed in the output of 'sip show settings', otherwise you can run into problems where the phone thinks that the subscription is valid for longer than Asterisk does. -- j. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Building an RPM from Asterisk 1.4
On 9/20/07, Marcus Franke [EMAIL PROTECTED] wrote: Do you have any examples for these spec files? I found a repository for installing Asterisk on Centos, but it took a while before I discovered it. Ok, just checked the link its for RHEL, but as Centos is just recompiled this won't matter. Same situation for Ubuntu using the debian package format, but I have not found a repository so far and Ubuntu delivers just the old 1.2 release. :) www.atrpms.net has pretty solid RPMs, and you can grab the SRPMS in order to get the spec file (either install the SRPM or use rpm2cpio to convert the package and extract the specfile manually). This is where I get my libpri and zaptel RPMs from (though I still build * from source, as the RPM compilation options they use are not to my liking). There is a book called Maximum RPM. The dead tree version is now pretty out of date with respect to the latest version of RPM (though still a good introduction if you've never built a package). I believe there was a slightly more up-to-date online version, but it still had some gaps the last time I looked. The best way to learn seems to be to examine good examples and then build your own package using their techniques. -- j. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Horrible problem - calls losing sound
On Thu, Sep 20, 2007 at 02:59:49PM +0200, John Hughes wrote: Yeah, I was being lazy, just using the asterisk from the Debian repositories. Bristuff includes its own ancient version of chan_capi. Debian removes it. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
On Thu, 20 Sep 2007 02:15:11 am Scott Moseman wrote: I'm getting frustrated simply trying to get this g729 working. For what it is worth, I had a similar issue to you, and managed to get g729 working by installing the binary files from http://asterisk.hosting.lv ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
I'm trying a simple Echo test and it's failing for g729... exten = 1267,1,Answer() exten = 1267,2,Echo() Test #1 (failure) gateway33 codecs g729a, g729b [gateway33] type=friend host=gateway33 context=default-inbound disallow=all allow=g729 gateway33 INVITE = g729b Asterisk 200 OK = no media Asterisk sends (one way) g711u RTP Test #2 (failure) gateway33 codecs g729a [gateway33] type=friend host=gateway33 context=default-inbound disallow=all allow=g729 gateway33 INVITE = g729a Asterisk 200 OK = no media gateway33 ends the session gateway33 INVITE = g729a Asterisk 200 OK = no media gateway33 ends the session ... Test #3 (success) gateway33 codec g729a, g729b, g711u [gateway33] type=friend host=gateway33 context=default-inbound disallow=all allow=ulaw allow=g729 gateway33 INVITE = g729b, g711u Asterisk 200 OK = g711u Asterisk sends/receives g711u RTP Does any of this point to a specific problem? I even have a licensed g729 channel. CLI show g729 0/0 encoders/decoders of 1 licensed channels are currently in use What information can I provide to help troubleshoot? This is making no sense. When I setup my desk phone to use G729 and make the test call directly (bypassing the gateway), the call completes fine and media is sent using G729 successfully. I'm not sure why it would work any differently from a Cisco gateway? The only difference that I'm aware of is that my phone (Polycom 430) seemed to ask for G729, while the gateway was either G729a or G729b specifically. In the instance of my phone, Asterisk came back with G729a in the 200 OK message. Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Building an RPM from Asterisk 1.4
On Thu, 2007-09-20 at 09:01 -0400, James FitzGibbon wrote: www.atrpms.net has pretty solid RPMs, and you can grab the SRPMS in order to get the spec file (either install the SRPM or use rpm2cpio to convert the package and extract the specfile manually). I've looked at a lot of Asterisk spec files (including the ones at atrpms.net, and the ones at http://www.laimbock.com/asterisk/), but so far my favorites are the ones done by Jeff C. Ollie at http://repo.ocjtech.us/misc/fedora/7/SRPMS/. They're about the most thorough I've seen. There is a book called Maximum RPM. The dead tree version is now pretty out of date with respect to the latest version of RPM (though still a good introduction if you've never built a package). The Red Hat RPM Guide is online at http://docs.fedoraproject.org/drafts/rpm-guide-en/, and it's fairly good. I also highly recommend the guide from Guru Labs, available at http://www.gurulabs.com/GURULABS-RPM-LAB/GURULABS-RPM-GUIDE-v1.0.PDF Personally, I agree with the original poster... the spec file that's currently in the Asterisk source is less than useful, and should probably be replaced. -- Jared Smith Community Relations Manager Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone RTP Session Start-up Delay
Thanks for the reply. I actually found the problem by inserting print statements in the code and checking which operations took the most time. I found that one line was taking a long time to run. The line was the following RTPManager.addtarget(destAddress) After googling this for a while, I found a solution on a JMF site which recommended a minor change in the way that the destAddress is obtained. After making this change the delay was removed. Thanks, Denis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James FitzGibbon Sent: Wednesday, September 19, 2007 4:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Softphone RTP Session Start-up Delay On 9/19/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: here because we are actually specifying the IP Address of the Asterisk server, but I am willing to try anything to fix this problem. The two user pc's are setup on workgroups, so I do not believe that there is a domain available that can be entered in the hosts file. Could the DNS still be the issue? If not, would anyone be able to suggest any other possible problems that may be causing this delay. It wasn't the same magnitude (more like 4 seconds for me), but I had an issue where the default eyeBeam (the commercial version of X-Lite) install was imposing a delay when a call first came in while it attempted to contact a non-existent STUN server. When I removed the STUN server setting, call setup was immediate. Might be worth looking at. Have you done a trace on the PC where the softphone is running (without a filter) to see what network packets are flying at the time the call setup happens? -- j. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outgoing SIP packets out of order?
Hello, I've been looking at some SIP packet dumps captured with tcpdump on the PBX itself, and analyzed with Wireshark 0.99.4. I'm noticing something strange, at least to me. All of the SIP packets going out from our Asterisk PBX to either of our 2 VoIP providers are consistently 50% out of order. In addition, if I use Wireshark's voip call player, the outgoing side of the call stutters and is delayed compared to the incoming side of the call. Is this normal? Why would the PBX be sending packets already out of order? Thanks! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY 14624 Office: 888-865-0065 Ext. 202 Mobile: (585) 721-8679 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with Asterisk behind a firewall
Did you set externip= ? - Original Message - From: Christian [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 12, 2007 3:23 PM Subject: [asterisk-users] Problems with Asterisk behind a firewall Hi all, I have set up Asterisk and I am able to register with my SIP provider and receive calls. When I try to register with Asterisk from outside I can place calls but tthe other person can't hear me. Have opened port 5060 UDP as well as port 1 to 2 UDP. Any ideas? Thanks, Christian ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] The device state is still 'Not in Use' ... check UPGRADE.txt
Or, in full: [Sep 20 17:11:26] WARNING[18373]: app_queue.c:2705 try_calling: The device state of this queue member, SIP/612, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. So, what do I check in UPGRADE.txt? This is with Asterisk 1.4.11 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The device state is still 'Not in Use' ... check UPGRADE.txt
John Hughes wrote: Or, in full: [Sep 20 17:11:26] WARNING[18373]: app_queue.c:2705 try_calling: The device state of this queue member, SIP/612, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. So, what do I check in UPGRADE.txt? This is with Asterisk 1.4.11 Ah, I'm so dumb - it's this bit: * Queues depend on the channel driver reporting the proper state for each member of the queue. To get proper signalling on queue members that use the SIP channel driver, you need to enable a call limit (could be set to a high value so it is not put into action) and also make sure that both inbound and outbound calls are accounted for. Example: [general] limitonpeer = yes [peername] type=friend call-limit=10 Could have sworn I had the call-limit, but in fact it was missing for the extension I was testing from! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Announcing: Click-to-Call with VIDEO
Dear All: Just as the name suggests, and evolving from regular Click-to-Call, Click-to-Call WITH VIDEO provides web sites with the ability to engage their visitors with a live video agent (plus the phone call). All with just a click of a button placed on the customer's web site. Please visit us at www.videoreps.net. The video technology integrates with Asterisk, and it also allows the agent to video-conference up to 10 callers at the same time, making it an ideal tool for online seminars, demos, shows, tech support, etc, etc. Click-to-call WITH VIDEO can be used in: 1) Customer service mode (web click-me buttons) 2) Conference mode (meetings of up to 10 participants) 3) Video-Call mode (enter user destination number and video-connect to him/her) It is an ideal add-on for pbx vendors who want to add video services to their pbxs. Best of all, it is free for now!! (video only)... Here is the deal: Free for one month, no commitments. Try, test it, call me. After a month you can decide if you want to keep any of our plans: Video-Call starts at $12.95 per month, and click-to-call with video at $29.95. And yes, there is 20% monthly commission if you include the service in the pbxs you are selling. Here is how it works: Register online and you will receive an email with the confirmation link. With your new username you will be able to enter the operator's work area. Please fill your profile with your web site's ip (so that the buttons work), your destination phone (so that the phone call works) and a meeting room username (so that the videocall works). Go to the demo buttons page and click on anyone you like to download the button and kit with a couple lines of php code you need to place the button in your web page. Next, assuming you have a camera connected, click on GoOnLine and you will be online!! (don't eat at your desk!). I appreciate your kind input on this web site. I am at my desk most of the time (logged in as a video agent), and I will be glad to be available when you click on our web site button, or when you enter my number 19176135931 on the meeting room for quick-connect videophone. However, hyper as I am, I would prefer to coordinate a demo time with anyone who wishes a demo session. Best Regards Christian Savinovich VideoReps.net Note: Being an ActiveX component, please use internet explorer. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI extension states
Here is what I use.
sub devstate2str($)
{
#func name stolen directly from asterisk
#takes int devstate and returns string val
my $ids = shift;
my $devstatestring = {};
$devstatestring-{0} = Unknown; #0 AST_DEVICE_UNKNOWN | Valid,
but unknown state
$devstatestring-{1} = Not in use; #1 AST_DEVICE_NOT_INUSE | Not used
$devstatestring-{2} = In use; #2 AST_DEVICE IN USE | In use
$devstatestring-{3} = Busy;#3 AST_DEVICE_BUSY | Busy
$devstatestring-{4} = Invalid; #4 AST_DEVICE_INVALID | Invalid
- not known to Asterisk
$devstatestring-{5} = Unavailable; #5 AST_DEVICE_UNAVAILABLE |
Unavailable (not registred)
$devstatestring-{6} = Ringing; #6 AST_DEVICE_RINGING | Ring,
ring, ring
return $devstatestring-{$ids};
}
LIke the comment says, I stole it direct from the source. A code of 16
may be a typo in which case you have found a bug.
Philipp Kempgen wrote:
Hi,
Is there a list of all the extension states as sent by the
manager interface? (I know I could look them up in the source
but that involves some backtracing.)
The ones I know are:
-1: no hint for the extension
0: registered idle
1: busy
4: unreachable, not registered
8: ringing
I've recently seen 16 (== hold?) but can't find that value
documented anywhere.
Regards,
Philipp Kempgen
--
Thank you and have a wonderful day,
Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI extension states
Oh one other note, when asking questions such as this, it is really wise to include which version # you are using. Philipp Kempgen wrote: Hi, Is there a list of all the extension states as sent by the manager interface? (I know I could look them up in the source but that involves some backtracing.) The ones I know are: -1: no hint for the extension 0: registered idle 1: busy 4: unreachable, not registered 8: ringing I've recently seen 16 (== hold?) but can't find that value documented anywhere. Regards, Philipp Kempgen -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The device state is still 'Not in Use' ... check UPGRADE.txt
John Hughes wrote: Or, in full: [Sep 20 17:11:26] WARNING[18373]: app_queue.c:2705 try_calling: The device state of this queue member, SIP/612, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. So, what do I check in UPGRADE.txt? This is with Asterisk 1.4.11 Here's what you're looking for: * Queues depend on the channel driver reporting the proper state for each member of the queue. To get proper signalling on queue members that use the SIP channel driver, you need to enable a call limit (could be set to a high value so it is not put into action) and also make sure that both inbound and outbound calls are accounted for. Example: [general] limitonpeer = yes [peername] type=friend call-limit=10 Mark Michelson ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
On 9/20/07, Luke Groeneveld [EMAIL PROTECTED] wrote: I'm getting frustrated simply trying to get this g729 working. For what it is worth, I had a similar issue to you, and managed to get g729 working by installing the binary files from http://asterisk.hosting.lv Thanks for the suggestion. Looks like I'm having the same problem, though. What's odd is that I can make phone to phone G729 calls through Asterisk, but G729 calls from my gateway do not work. Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with asterisk-perl-0.08 and Asterisk = 1.2.20
Tzafrir Cohen a écrit : On Wed, Sep 19, 2007 at 12:25:35PM +0200, Benoît Mérouze wrote: Is there any reason this can be fixed in the asterisk-perl-0.10 (not yet included in Trixbox)? Or is this more an issue from Asterisk (since Asterisk 1.2.19 or 1.2.20)? Why not give it a shot? Install asterisk-perl's modules to a non-standard path and add to your script a 'use' directive to use that path first. I've tried asterisk-perl-0.10 on a development server. And the result is the same. Hence I guess something has changed between Asterisk-1.2.18 and later revisions that no more works with asterisk-perl when get_variable is called (calling Asterisk AGI command GET VARIABLE). I've also discovered the intersting function get_full_variable in asterisk-perl-0.10. But the issue is the same with Asterisk 1.2.18. Regards, Benoit -- Benoît Mérouze - Telecom Software Developer - IPercom [EMAIL PROTECTED] Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. Benjamin Franklin ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue Question
I'm curious if anyone has implemented the following: Need to setup an on-call queue, that activates after 5PM and de-activates at 8AM, also that activates/deactivates on demand(I'm thinking a feature code here). The agents need to log in via cell phones, and when calls come in from outside to the asterisk system, it'll need to call the cell phone agents that are active. I'm thinking that it's a simple SQL query, to update the agents status and number, and that asterisk will do a lookup and append that to the ZAP channel to dial, but interested in any logic someone might be able to come up with for the dialplan. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI extension states
Anthony Francis wrote:
Here is what I use.
sub devstate2str($)
{
#func name stolen directly from asterisk
#takes int devstate and returns string val
my $ids = shift;
my $devstatestring = {};
$devstatestring-{0} = Unknown; #0 AST_DEVICE_UNKNOWN | Valid,
but unknown state
$devstatestring-{1} = Not in use; #1 AST_DEVICE_NOT_INUSE | Not used
$devstatestring-{2} = In use; #2 AST_DEVICE IN USE | In use
$devstatestring-{3} = Busy;#3 AST_DEVICE_BUSY | Busy
$devstatestring-{4} = Invalid; #4 AST_DEVICE_INVALID | Invalid
- not known to Asterisk
$devstatestring-{5} = Unavailable; #5 AST_DEVICE_UNAVAILABLE |
Unavailable (not registred)
$devstatestring-{6} = Ringing; #6 AST_DEVICE_RINGING | Ring,
ring, ring
return $devstatestring-{$ids};
}
LIke the comment says, I stole it direct from the source. A code of 16
may be a typo in which case you have found a bug.
Philipp Kempgen wrote:
Hi,
Is there a list of all the extension states as sent by the
manager interface? (I know I could look them up in the source
but that involves some backtracing.)
The ones I know are:
-1: no hint for the extension
0: registered idle
1: busy
4: unreachable, not registered
8: ringing
I've recently seen 16 (== hold?) but can't find that value
documented anywhere.
Thanks, but I think device states are different from extension states.
(Please correct me If I'm mistaken.)
Device states from devicestate.h:
---cut---
/*! Device is valid but channel didn't know state */
#define AST_DEVICE_UNKNOWN 0
/*! Device is not used */
#define AST_DEVICE_NOT_INUSE1
/*! Device is in use */
#define AST_DEVICE_INUSE2
/*! Device is busy */
#define AST_DEVICE_BUSY 3
/*! Device is invalid */
#define AST_DEVICE_INVALID 4
/*! Device is unavailable */
#define AST_DEVICE_UNAVAILABLE 5
/*! Device is ringing */
#define AST_DEVICE_RINGING 6
/*! Device is ringing *and* in use */
#define AST_DEVICE_RINGINUSE7
/*! Device is on hold */
#define AST_DEVICE_ONHOLD 8
---cut---
I was looking for the extension states as sent by the manager interface.
Extension states from pbx.h:
---cut---
/*! \brief Extension states */
enum ast_extension_states {
AST_EXTENSION_REMOVED = -2, /*! Extension removed */
AST_EXTENSION_DEACTIVATED = -1, /*! Extension hint removed */
AST_EXTENSION_NOT_INUSE = 0,/*! No device INUSE or BUSY */
AST_EXTENSION_INUSE = 1 0, /*! One or more devices INUSE */
AST_EXTENSION_BUSY = 1 1,/*! All devices BUSY */
AST_EXTENSION_UNAVAILABLE = 1 2, /*! All devices
UNAVAILABLE/UNREGISTERED */
AST_EXTENSION_RINGING = 1 3, /*! All devices RINGING */
AST_EXTENSION_ONHOLD = 1 4, /*! All devices ONHOLD */
};
---cut---
14 == 16, so probably this is the value I observed.
But there some mangling occurs in ast_extension_state2() in pbx.c like
if (inuse ring)
return (AST_EXTENSION_INUSE | AST_EXTENSION_RINGING);
which would return
10 | 13 == 1|8 == 9
So the extension state is some kind of a bit mask.
I think I have found what I was looking for. :)
Thanks,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? - http://www.das-asterisk-buch.de
My pick of the month: rfc 2822 3.6.5
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI extension states
Anthony Francis wrote: Oh one other note, when asking questions such as this, it is really wise to include which version # you are using. Right. Sorry. 1.4.11 (for the archives) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
I am trying to connect two machines to each other with an T1 crossover cable. The first machine has two TE120P cards - one connecting to the telco on an ISDN PRI. The second to a crossover T1 cable to a second machine which has one TE120P card. Telco -cA- Machine1 -cB- Machine2 Machine1: Two TE120P cards Machine2: One TE120P card cA: Standard T1 Cable cB: Crossover T1 Cable Configuration files are included at the end of this message. I have used both 'cat /proc/interupts' and 'lspci -vb' to verify that the cards do not have IRQ conflicts. Machine2 can be plugged into the telco pri and works fine. Machine1 works on the telco PRI so long as I have removed the configuration for the second span (the one on the crossover). I can leave the card for the second span in - so long as it is not configured. Machine 1 CLI Notices If both cards are configured and connected. Zttool reliably reports no alarms. Asterisk appears to start without errors at verbosity three. Very quickly after it starts Asterisk starts reporting WARNING[28899] chan_zap.c:6668 handle_init_event: Detected alarm on channel NN: Red Alarm WARNING[28899] chan_zap.c:1464 zt_disable_ec: Unable to disable echo cancellation on channel NN It will output these for all the channels on the second span (NN ranges from 25 to 47). It then will report NOTICE[28898] chan_zap.c:8460 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 2 WARNING[28898] chan_zap.c:2393 pri_find_dchan: No D-channels available! Using primary channel 48 as D-channel anyway! This is immediately followed by a notice that the alarms were cleared on each of the channels of span 2, including the data-channel. NOTICE [29899]: chan_zap.c:6661 handle_init_event: Alarm cleared on channel NN NOTICE [29898]: chan_zap.c:8460 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 2 This is then followed by the first error ERROR [29898]: chan_zap.c:8174 zt_pri_error: !! Got S-frame while link down I then get constant notices that Primary D-Channel on span 2 up. This will be periodically broken up by a repeat of the warnings, notices and errors I describe above. However, now the problems occur for all of the channels of both spans. Machine 1 CLI Notices Machine 1 experiences almost the same behavior on its span. The only differance I am noticing is that instead of the S-frame error I get the following notice: chan_zap.c:8457 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Zap Restart Fails On either machine if I attempt a 'zap restart' I receive WARNING[30686]: chan_zap.c:903 zt_open: Unable to specify channel 1: Device or resource busy ERROR[30686]: chan_zap.c:7160 mkintf: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 ERROR[30686]: chan_zap.c:10467 build_channels: Unable to register channel '1-23' WARNING[30686]: chan_zap.c:9764 zap_restart: Reload channels from zap config failed! I have not attempted to connect two Asterisk boxes though a T1 crossover before so I am stumped. I have included my zaptel.conf and zapata.conf files below. I will certainly appreciate any help you can give. Thanks, -Brian Machine1 = zaptel.conf --- defaultzone=us loadzone=us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 zapata.conf --- [trunkgroups] [channels] group=1 context=fromtelco signalling=pri_cpe switchtype=national channel=1-23 group=1 context=frommachine2 signalling=pri_net switchtype=national channel=25-47 Machine2 = zaptel.conf --- defaultzone=us loadzone=us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf --- [trunkgroups] [channels] group=1 context=frommachine1 signalling=pri_cpe switchtype=national channel=1-23 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI extension states
Philipp Kempgen wrote:
Anthony Francis wrote:
Here is what I use.
sub devstate2str($)
{
#func name stolen directly from asterisk
#takes int devstate and returns string val
my $ids = shift;
my $devstatestring = {};
$devstatestring-{0} = Unknown; #0 AST_DEVICE_UNKNOWN | Valid,
but unknown state
$devstatestring-{1} = Not in use; #1 AST_DEVICE_NOT_INUSE | Not used
$devstatestring-{2} = In use; #2 AST_DEVICE IN USE | In use
$devstatestring-{3} = Busy;#3 AST_DEVICE_BUSY | Busy
$devstatestring-{4} = Invalid; #4 AST_DEVICE_INVALID | Invalid
- not known to Asterisk
$devstatestring-{5} = Unavailable; #5 AST_DEVICE_UNAVAILABLE |
Unavailable (not registred)
$devstatestring-{6} = Ringing; #6 AST_DEVICE_RINGING | Ring,
ring, ring
return $devstatestring-{$ids};
}
LIke the comment says, I stole it direct from the source. A code of 16
may be a typo in which case you have found a bug.
Philipp Kempgen wrote:
Hi,
Is there a list of all the extension states as sent by the
manager interface? (I know I could look them up in the source
but that involves some backtracing.)
The ones I know are:
-1: no hint for the extension
0: registered idle
1: busy
4: unreachable, not registered
8: ringing
I've recently seen 16 (== hold?) but can't find that value
documented anywhere.
Thanks, but I think device states are different from extension states.
(Please correct me If I'm mistaken.)
Device states from devicestate.h:
---cut---
/*! Device is valid but channel didn't know state */
#define AST_DEVICE_UNKNOWN0
/*! Device is not used */
#define AST_DEVICE_NOT_INUSE 1
/*! Device is in use */
#define AST_DEVICE_INUSE 2
/*! Device is busy */
#define AST_DEVICE_BUSY 3
/*! Device is invalid */
#define AST_DEVICE_INVALID4
/*! Device is unavailable */
#define AST_DEVICE_UNAVAILABLE5
/*! Device is ringing */
#define AST_DEVICE_RINGING6
/*! Device is ringing *and* in use */
#define AST_DEVICE_RINGINUSE 7
/*! Device is on hold */
#define AST_DEVICE_ONHOLD 8
---cut---
I was looking for the extension states as sent by the manager interface.
Extension states from pbx.h:
---cut---
/*! \brief Extension states */
enum ast_extension_states {
AST_EXTENSION_REMOVED = -2, /*! Extension removed */
AST_EXTENSION_DEACTIVATED = -1, /*! Extension hint removed */
AST_EXTENSION_NOT_INUSE = 0,/*! No device INUSE or BUSY */
AST_EXTENSION_INUSE = 1 0, /*! One or more devices INUSE */
AST_EXTENSION_BUSY = 1 1,/*! All devices BUSY */
AST_EXTENSION_UNAVAILABLE = 1 2, /*! All devices
UNAVAILABLE/UNREGISTERED */
AST_EXTENSION_RINGING = 1 3, /*! All devices RINGING */
AST_EXTENSION_ONHOLD = 1 4, /*! All devices ONHOLD */
};
---cut---
14 == 16, so probably this is the value I observed.
But there some mangling occurs in ast_extension_state2() in pbx.c like
if (inuse ring)
return (AST_EXTENSION_INUSE | AST_EXTENSION_RINGING);
which would return
10 | 13 == 1|8 == 9
So the extension state is some kind of a bit mask.
I think I have found what I was looking for. :)
Thanks,
Philipp Kempgen
Umm, that is the code I use in my manager interface monitoring software,
and it works. of course I am using 1.2.18 and perhaps you are using a
different version.
--
Thank you and have a wonderful day,
Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI extension states
Philipp Kempgen wrote: Anthony Francis wrote: Oh one other note, when asking questions such as this, it is really wise to include which version # you are using. Right. Sorry. 1.4.11 (for the archives) Regards, Philipp Kempgen That is what I thought, makes what I said invalid for you, but valid in 1.2 -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Samsung Sprint CDMAoIP
The device is a femtocell device; I would bet that they keep it in a format that works with their existing equipment, rather than use SIP. The device is also licensed for a specific frequency that is owned by the carrier, you wouldn't be able to use this device for any other purpose without their permission. The cell carriers are trying to get people excited about femotcell technology, so they can shift the cellular infrastructure costs off onto their customers. The traditional model is that the cell providers pay property and tower owners rent when they site an antenna. The cell providers don't even offer you a cheaper rate when you use the infrastructure you paid for. Notice how the device supports up to three simultaneous calls? You're even paying them to provide a better signal for your neighbors. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, September 19, 2007 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] OT: Samsung Sprint CDMAoIP http://gizmodo.com/gadgets/cellphones/sprintsamsung-instant-cell+to+wi+fi- box-is-official-named-airave-300451.php The above is quite interesting, it would be interesting to see if it uses sip, which I have no reason to believe otherwise, and if it does, can it be hacked to talk to Asteirsk? In which case one could have a very good extension to asterisk using any Sprint Cell phone, or maybe even any CDMA (Verizon) cell phone as well. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Announcing: Click-to-Call with VIDEO ***SPAM***
Can we please block these clowns? It appears they are incapable of learning. C. Savinovich wrote: Dear All: Just as the name suggests, and evolving from regular Click-to-Call, Click-to-Call WITH VIDEO provides web sites with the ability to engage their visitors with a live video agent (plus the phone call). All with just a click of a button placed on the customer's web site. Please visit us at www.videoreps.net. The video technology integrates with Asterisk, and it also allows the agent to video-conference up to 10 callers at the same time, making it an ideal tool for online seminars, demos, shows, tech support, etc, etc. Click-to-call WITH VIDEO can be used in: 1) Customer service mode (web click-me buttons) 2) Conference mode (meetings of up to 10 participants) 3) Video-Call mode (enter user destination number and video-connect to him/her) It is an ideal add-on for pbx vendors who want to add video services to their pbxs. Best of all, it is free for now!! (video only)... Here is the deal: Free for one month, no commitments. Try, test it, call me. After a month you can decide if you want to keep any of our plans: Video-Call starts at $12.95 per month, and click-to-call with video at $29.95. And yes, there is 20% monthly commission if you include the service in the pbxs you are selling. Here is how it works: Register online and you will receive an email with the confirmation link. With your new username you will be able to enter the operator's work area. Please fill your profile with your web site's ip (so that the buttons work), your destination phone (so that the phone call works) and a meeting room username (so that the videocall works). Go to the demo buttons page and click on anyone you like to download the button and kit with a couple lines of php code you need to place the button in your web page. Next, assuming you have a camera connected, click on GoOnLine and you will be online!! (don't eat at your desk!). I appreciate your kind input on this web site. I am at my desk most of the time (logged in as a video agent), and I will be glad to be available when you click on our web site button, or when you enter my number 19176135931 on the meeting room for quick-connect videophone. However, hyper as I am, I would prefer to coordinate a demo time with anyone who wishes a demo session. Best Regards Christian Savinovich VideoReps.net Note: Being an ActiveX component, please use internet explorer. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hfcmulti and B410P Digium Card
voip crazy wrote: Hello all, I am getting the following error in /var/log/syslog. I have got 2 B410P cards in this box. Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=0153, z2=00d3) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0013, z2=0012) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=0053, z2=0153) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0093, z2=0092) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=00d3, z2=0053) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0113, z2=0112) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=0153, z2=00d3) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0013, z2=0012) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=0053, z2=0153) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(2) reading 15 bytes (z1=004e, z2=0040) HDLC COMPLETE (f1=3, f2=2) got=15 Sep 19 17:13:31 localhost kernel: 02 01 06 06 08 01 63 5a 08 02 80 90 Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0093, z2=0092) sending 128 of 128 bytes TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(2) has 383 bytes space left (z1=015d, z2=015d) sending 4 of 4 bytes HDLC I left untouched the /etc/init.d/misdn-init script to load the default values. Is needed the hfcmulti modules with this kind of cards? What is the menaing of this errors? Are something missconfigured? Unless you are having some sort of problem other than this, than I think that this is just standard debug output, which you can disable if you set the debug option in /etc/misdn-init.conf to 0. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX Java Softphone?
Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is softswitch
Alex Balashov wrote: On Wed, 19 Sep 2007, Anthony Francis wrote: IMHO asterisk is a softswitch, it may not be a very high capacity one (right now) but it can be and if you don't mind splitting your physical trunk calls over multiple machines it works very well as a call routing engine, you just need to have carefully designed plans. It is far to easy to create call routing loops, but if you don't know what you are doing with a real telephony switch you can do the same. No SS7/ISUP support (and no TCAP, which is required for LNP and LIDB and traditional CNAM), poor/incomplete IMT support, can't take more than a few T1s per host - if that. No GR.303 support. Actually, I have been working on an SS7 stack for asterisk called libss7. SS7 support is already in trunk, and should be in the next stable release of Asterisk. Right now it only does ISUP/MTP3/MTP2, but with some work an effort, SCCP/TCAP/LNP support could be implemented as well. Asterisk has had GR.303 support for a while, though I don't think it's asymmetric (it only supports one particular function of it, or something like that IIRC). -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is softswitch
On Thu, 20 Sep 2007, Matthew Fredrickson wrote: Actually, I have been working on an SS7 stack for asterisk called libss7. SS7 support is already in trunk, and should be in the next stable release of Asterisk. Right now it only does ISUP/MTP3/MTP2 Look forward to it! That will certainly confer an entirely new brand of utility upon Asterisk. but with some work an effort, SCCP/TCAP/LNP support could be implemented as well. If the work and effort is something Digium decides to invest, it's certainly going to revolutionise what one can do with Asterisk. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO
He changed the title of his response. Your post remains intact on the list in its original form. In the interest of letting others decide, I think it was a spammy post as well. Others have decided... Hence the title of the list Asterisk Users Mailing List - Non-Commercial Discussion. They decided it a long time ago. When you included pricing, your email became commercial, an advertisement, and spam. Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Savinovich Sent: Thursday, September 20, 2007 4:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO Please don't change the title of my post. It is disrespectful. One thing is to give your opinion about its content, and another to be self appointed editor of this forum. We posted in this forum because it is a contribution to the asterisk community, and because it is free for a month, and maybe even longer if the community so demands it. If you agree or disagree with it fine, but let others decide. They know spam when they see it. Thank you. C. Savinovich VideoReps.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Thursday, September 20, 2007 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO Am Donnerstag, den 20.09.2007, 11:30 -0700 schrieb C. Savinovich: Dear All: Just as the name suggests, and evolving from regular Click-to-Call, Click-to-Call WITH VIDEO provides web sites with the ability to engage their visitors with a live video agent (plus the phone call). All with just a click of a button placed on the customer's web site. Please visit us at www.videoreps.net. When I read this, I thougt: Wow, here comes a nice, free, open, interesting software. Best of all, it is free for now!! (video only)... Here is the deal: Free for one month, no commitments. Try, test it, call me. After a month you can decide if you want to keep any of our plans: Video-Call starts at $12.95 per month, and click-to-call with video at $29.95. And yes, there is 20% monthly commission if you include the service in the pbxs you are selling. Well, not free at all. Note: Being an ActiveX component, please use internet explorer. And not having too much to do with open either, I guess. I would have called you to personally tell you that your mail was misplaced (there is some kind of asterisk-biz list, and I do not read it for a purpose), but I do not use the integrated exploder for lack of the necessary obfuscation system on my work machine. Please do not send commercials, ads and product information to this list. It might very well be considered SPAM. Just and only because _some_ readers might be interested there is no legitimation for sending it (else every pen15-en1argm3nt _might_ trigger interest at some readers). Thanks Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ***SPAM*** Announcing: Click-to -Call with VIDEO
On Thursday 20 September 2007, C. Savinovich wrote: We posted in this forum because it is a contribution to the asterisk community, and because it is free for a month, and maybe even longer if the community so demands it. If you agree or disagree with it fine, but let others decide. They know spam when they see it. Thank you. I am one of the longest running developers of Asterisk and I agree with Anselm's assessment. Please post announcements like this in the future to the asterisk-biz list, as that list is for business discussions. This list is for users of Asterisk, not for commercial announcements. -- Tilghman ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
On Thu, 2007-09-20 at 13:20 -0400, Brian Alexander wrote: Machine 1 experiences almost the same behavior on its span. The only differance I am noticing is that instead of the S-frame error I get the following notice: chan_zap.c:8457 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 I'd look at your wiring, as an HDLC error like that is usually an indication of some type of a problem at the physical layer. -- Jared Smith Community Relations Manager Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Java Softphone?
I use click2call. http://www.geocities.com/babarnazmi/index2.htm It is an activex control though. -- -- Steven http://www.glimasoutheast.org Matthew Rubenstein [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Samsung Sprint CDMAoIP
On 9/20/07, Eric Chamberlain [EMAIL PROTECTED] wrote: The device is a femtocell device; I would bet that they keep it in a format that works with their existing equipment, rather than use SIP. The device is also licensed for a specific frequency that is owned by the carrier, you wouldn't be able to use this device for any other purpose without their permission. The cell carriers are trying to get people excited about femotcell technology, so they can shift the cellular infrastructure costs off onto their customers. The traditional model is that the cell providers pay property and tower owners rent when they site an antenna. AFAIK, the calls are free when you use it thru that device. Sprint however charges $15 a month per phone or $30 for family plan. While I agree that sprint should pay me for this, it's not as bad. T-mobile on the other hand, does the same thing with wifi enabled phones, it doesn't cost extra, and is completely free. The cell providers don't even offer you a cheaper rate when you use the infrastructure you paid for. Notice how the device supports up to three simultaneous calls? You're even paying them to provide a better signal for your neighbors. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, September 19, 2007 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] OT: Samsung Sprint CDMAoIP http://gizmodo.com/gadgets/cellphones/sprintsamsung-instant-cell+to+wi+fi- box-is-official-named-airave-300451.php The above is quite interesting, it would be interesting to see if it uses sip, which I have no reason to believe otherwise, and if it does, can it be hacked to talk to Asteirsk? In which case one could have a very good extension to asterisk using any Sprint Cell phone, or maybe even any CDMA (Verizon) cell phone as well. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Java Softphone?
Matthew Rubenstein wrote: Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. Mexuar's Coraletta is nice, but isn't GPL. http://www.mexuar.com/products_sdk.shtml Mike Clark ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO
I was interested in it - commercial or otherwise.but only because I used to work for the competition. Commercial or otherwise it looks like a very cool technology and something I'd be interested in - but only as a one of purchase price rather than an ASP. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Martin Smith Sent: Thursday, 20 September 2007 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO He changed the title of his response. Your post remains intact on the list in its original form. In the interest of letting others decide, I think it was a spammy post as well. Others have decided... Hence the title of the list Asterisk Users Mailing List - Non-Commercial Discussion. They decided it a long time ago. When you included pricing, your email became commercial, an advertisement, and spam. Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Savinovich Sent: Thursday, September 20, 2007 4:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO Please don't change the title of my post. It is disrespectful. One thing is to give your opinion about its content, and another to be self appointed editor of this forum. We posted in this forum because it is a contribution to the asterisk community, and because it is free for a month, and maybe even longer if the community so demands it. If you agree or disagree with it fine, but let others decide. They know spam when they see it. Thank you. C. Savinovich VideoReps.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Thursday, September 20, 2007 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO Am Donnerstag, den 20.09.2007, 11:30 -0700 schrieb C. Savinovich: Dear All: Just as the name suggests, and evolving from regular Click-to-Call, Click-to-Call WITH VIDEO provides web sites with the ability to engage their visitors with a live video agent (plus the phone call). All with just a click of a button placed on the customer's web site. Please visit us at www.videoreps.net. When I read this, I thougt: Wow, here comes a nice, free, open, interesting software. Best of all, it is free for now!! (video only)... Here is the deal: Free for one month, no commitments. Try, test it, call me. After a month you can decide if you want to keep any of our plans: Video-Call starts at $12.95 per month, and click-to-call with video at $29.95. And yes, there is 20% monthly commission if you include the service in the pbxs you are selling. Well, not free at all. Note: Being an ActiveX component, please use internet explorer. And not having too much to do with open either, I guess. I would have called you to personally tell you that your mail was misplaced (there is some kind of asterisk-biz list, and I do not read it for a purpose), but I do not use the integrated exploder for lack of the necessary obfuscation system on my work machine. Please do not send commercials, ads and product information to this list. It might very well be considered SPAM. Just and only because _some_ readers might be interested there is no legitimation for sending it (else every pen15-en1argm3nt _might_ trigger interest at some readers). Thanks Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or
Re: [asterisk-users] IAX Java Softphone?
Matthew Rubenstein a écrit : Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. Did you try JIAXClient ? http://www.hem.za.org/jiaxclient/ Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Java Softphone?
Steven, how reliable is that freeware? I tried it when it first came out but I couldn't get it to work. It didn't matter at the time as I was working for Mexuar at the time but now I don't have their service anymore I'd like to use it/something like it for my other consultancy services. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steven Sent: Thursday, 20 September 2007 2:12 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] IAX Java Softphone? I use click2call. http://www.geocities.com/babarnazmi/index2.htm It is an activex control though. -- -- Steven http://www.glimasoutheast.org Matthew Rubenstein [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Samsung Sprint CDMAoIP
C F wrote: AFAIK, the calls are free when you use it thru that device. Sprint however charges $15 a month per phone or $30 for family plan. While I agree that sprint should pay me for this, it's not as bad. T-mobile on the other hand, does the same thing with wifi enabled phones, it doesn't cost extra, and is completely free. If you're referring to T-Mobile's [EMAIL PROTECTED] service, it's actually $20 per month, per line on the account (unless it's changed very recently). As far as how it works on T-Mobile, I recently had some questions answered by them about that.. They use UMA over wifi, and it does automatic switching between the wifi and the gsm towers (ie; your call stays up). Quote from the tech I talked to: [EMAIL PROTECTED] does not use a VoIP protocol, as the voice data is transferred from the Internet directly to our UMA Gateway and then through our regular Mobile Switching Centers. Pretty interesting stuff. -- Jason Parker Digium ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO
snip Please don't change the title of my post. It is disrespectful. One thing is to give your opinion about its content, and another to be self appointed editor of this forum. We posted in this forum because it is a contribution to the asterisk community, and because it is free for a month, and maybe even longer if the community so demands it. If you agree or disagree with it fine, but let others decide. They know spam when they see it. Thank you. C. Savinovich VideoReps.net snip As one of the others I say it looks like spam to me too. I won't be trying your product anytime soon, partly because of the way the matter was handled. -Troy ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newcomer Question
Hallo Martin and Group! Thank you very much for your perfect introduction into asterisk. I managed to * get asterisk server running * configuring the internal numbers * registering to 2 sip gateways * outbound phoning to sipgate works perfect * outbound phoning to mujtelefon not yet tested The problem i am having now is, that i cant be reached by inbound phone calls from neither sipgate nor mujtelefon i used my mobile to call this numbers. sipgate tells me on the phone:Das Endgeraet ist fuer diesen Service nicht konfiguriert. bei mujtelefon kommt nur die mobilbox In asterisk cli i dont see anything about that. in the sipgate login page there is neither mentioned. If this is working, I intend to have nice music played for incoming calls until the phone call is accepted. What is very confusing for sipgate is, that my number(734365) is different from my user name(1734365). Can anybopdy check, if all settings are ok according to that ? Please find below my sip.conf, only the passwords are scrambled. If you directly reply to me, also reply to [EMAIL PROTECTED] because i am afraid missing your answer in the much traffic in that mailing list. Thank you very much! [general] bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm musicclass=default language=de dtmfmode=rfc2833 sipdebug=no register = 1734365:[EMAIL PROTECTED]:5060/00437201734365 register = 272048160:[EMAIL PROTECTED]:5060/00420272048160 [sipgateat] host=sipgate.at secret=NMMTNMKP username=1734365 fromuser=1734365 fromdomain=sipgate.at srvlookup=yes context=sipgateat-in canreinvite=no nat=no type=friend qualify=yes
Re: [asterisk-users] IAX Java Softphone?
On Thu, 2007-09-20 at 14:23 -0400, Mike Clark wrote: Matthew Rubenstein wrote: Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. Mexuar's Coraletta is nice, but isn't GPL. http://www.mexuar.com/products_sdk.shtml I'm using JIAXClient [1], it is GPL, uses IAX2 and works pretty excelent with gsm codec. [1] http://www.hem.za.org/jiaxclient/ Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Announcing: Click-to-Call with VIDEO
OK gentlemen, thank you very much. Best Regards C. Savinovich VideoReps.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Thursday, September 20, 2007 11:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO On Thursday 20 September 2007, C. Savinovich wrote: We posted in this forum because it is a contribution to the asterisk community, and because it is free for a month, and maybe even longer if the community so demands it. If you agree or disagree with it fine, but let others decide. They know spam when they see it. Thank you. I am one of the longest running developers of Asterisk and I agree with Anselm's assessment. Please post announcements like this in the future to the asterisk-biz list, as that list is for business discussions. This list is for users of Asterisk, not for commercial announcements. -- Tilghman ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Java Softphone?
As far as I know Jiaxclient is dead - the developer hasn't touched it in at least 18 months. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jean-Denis Girard Sent: Thursday, 20 September 2007 2:46 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Java Softphone? Matthew Rubenstein a écrit : Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. Did you try JIAXClient ? http://www.hem.za.org/jiaxclient/ Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Astricon Ride From Airport to Conf Hotel
Hi All, I'm arriving around noon in Phoenix on Tuesday the 25'Th and wouldn't mind sharing a cab or car service. I spoke with the hotel and the 'Super Shuttle' service can take 2-3 hours because the resort is the last stop on the route. A cab or car service will only take 30-40 minutes. If anyone is coming in around the same time and needs a ride, contact me off-list. Thanks. JR -- JR Richardson Engineering for the Masses ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outgoing SIP packets out of order?
Are you confident it's not a defect in Wireshark's RTP analyser? On Thu, 20 Sep 2007, Jason Martin wrote: Hello, I've been looking at some SIP packet dumps captured with tcpdump on the PBX itself, and analyzed with Wireshark 0.99.4. I'm noticing something strange, at least to me. All of the SIP packets going out from our Asterisk PBX to either of our 2 VoIP providers are consistently 50% out of order. In addition, if I use Wireshark's voip call player, the outgoing side of the call stutters and is delayed compared to the incoming side of the call. Is this normal? Why would the PBX be sending packets already out of order? Thanks! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY 14624 Office: 888-865-0065 Ext. 202 Mobile: (585) 721-8679 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Announcing: Click-to-Call with VIDEO
Dear Dean: It is a very cool technology indeed, and please, do not see me as your competition, but as a friend. I know you have a click-to-call product, and if there is any way I can be of help with providing the video technology for you, I will be glad to set it up for you. You are most welcomed to use videoreps for free until you establish how it can benefit you. C. Savinovich VideoReps -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Thursday, September 20, 2007 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [PHISH] Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO I was interested in it - commercial or otherwise.but only because I used to work for the competition. Commercial or otherwise it looks like a very cool technology and something I'd be interested in - but only as a one of purchase price rather than an ASP. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Martin Smith Sent: Thursday, 20 September 2007 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO He changed the title of his response. Your post remains intact on the list in its original form. In the interest of letting others decide, I think it was a spammy post as well. Others have decided... Hence the title of the list Asterisk Users Mailing List - Non-Commercial Discussion. They decided it a long time ago. When you included pricing, your email became commercial, an advertisement, and spam. Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Savinovich Sent: Thursday, September 20, 2007 4:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO Please don't change the title of my post. It is disrespectful. One thing is to give your opinion about its content, and another to be self appointed editor of this forum. We posted in this forum because it is a contribution to the asterisk community, and because it is free for a month, and maybe even longer if the community so demands it. If you agree or disagree with it fine, but let others decide. They know spam when they see it. Thank you. C. Savinovich VideoReps.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Thursday, September 20, 2007 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO Am Donnerstag, den 20.09.2007, 11:30 -0700 schrieb C. Savinovich: Dear All: Just as the name suggests, and evolving from regular Click-to-Call, Click-to-Call WITH VIDEO provides web sites with the ability to engage their visitors with a live video agent (plus the phone call). All with just a click of a button placed on the customer's web site. Please visit us at www.videoreps.net. When I read this, I thougt: Wow, here comes a nice, free, open, interesting software. Best of all, it is free for now!! (video only)... Here is the deal: Free for one month, no commitments. Try, test it, call me. After a month you can decide if you want to keep any of our plans: Video-Call starts at $12.95 per month, and click-to-call with video at $29.95. And yes, there is 20% monthly commission if you include the service in the pbxs you are selling. Well, not free at all. Note: Being an ActiveX component, please use internet explorer. And not having too much to do with open either, I guess. I would have called you to personally tell you that your mail was misplaced (there is some kind of asterisk-biz list, and I do not read it for a purpose), but I do not use the integrated exploder for lack of the necessary obfuscation system on my work machine. Please do not send commercials, ads and product information to this list. It might very well be considered SPAM. Just and only because _some_ readers might be interested there is no legitimation for sending it (else every pen15-en1argm3nt _might_ trigger interest at some readers). Thanks Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list
[asterisk-users] Polycom 330 + Asterisk, phone locks up. * key will do it
Hello all, We have an Asterisk server that has worked without issue for a while. Before, only Sipura and Polycom 500 series phones were used. Recently, we've added a few POE switches and 20 or so Polycom 330's. The 330's seem to lock up often. One easy way to do this is by hitting the * key. I looked thru the XML files for dialplan and what not, nothing stood out. I've tried the latest everyone-can-download (bootrom 3.2.3.0021 and firmware 2.1.2.0048). Thoughts? The 500 series phones (501) have no issues. Also, the phones log called ID information and make it easy for users to hit a button and call back missed calls. Except, it doesn't log a 1 in front of the number. I guess I could adjust the dialplan to assume 9 digit numbers need a 1 in front, but has anyone found a better way? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Announcing: Click-to-Call with VIDEO
I haven't been involved with Mexuar for about 4 months. In the middle of moving at the moment but will be in touch in about 2 weeks once things get back to normal. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C. Savinovich Sent: Thursday, 20 September 2007 6:26 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Announcing: Click-to-Call with VIDEO Dear Dean: It is a very cool technology indeed, and please, do not see me as your competition, but as a friend. I know you have a click-to-call product, and if there is any way I can be of help with providing the video technology for you, I will be glad to set it up for you. You are most welcomed to use videoreps for free until you establish how it can benefit you. C. Savinovich VideoReps -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Thursday, September 20, 2007 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [PHISH] Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO I was interested in it - commercial or otherwise.but only because I used to work for the competition. Commercial or otherwise it looks like a very cool technology and something I'd be interested in - but only as a one of purchase price rather than an ASP. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Martin Smith Sent: Thursday, 20 September 2007 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO He changed the title of his response. Your post remains intact on the list in its original form. In the interest of letting others decide, I think it was a spammy post as well. Others have decided... Hence the title of the list Asterisk Users Mailing List - Non-Commercial Discussion. They decided it a long time ago. When you included pricing, your email became commercial, an advertisement, and spam. Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Savinovich Sent: Thursday, September 20, 2007 4:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO Please don't change the title of my post. It is disrespectful. One thing is to give your opinion about its content, and another to be self appointed editor of this forum. We posted in this forum because it is a contribution to the asterisk community, and because it is free for a month, and maybe even longer if the community so demands it. If you agree or disagree with it fine, but let others decide. They know spam when they see it. Thank you. C. Savinovich VideoReps.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Thursday, September 20, 2007 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO Am Donnerstag, den 20.09.2007, 11:30 -0700 schrieb C. Savinovich: Dear All: Just as the name suggests, and evolving from regular Click-to-Call, Click-to-Call WITH VIDEO provides web sites with the ability to engage their visitors with a live video agent (plus the phone call). All with just a click of a button placed on the customer's web site. Please visit us at www.videoreps.net. When I read this, I thougt: Wow, here comes a nice, free, open, interesting software. Best of all, it is free for now!! (video only)... Here is the deal: Free for one month, no commitments. Try, test it, call me. After a month you can decide if you want to keep any of our plans: Video-Call starts at $12.95 per month, and click-to-call with video at $29.95. And yes, there is 20% monthly commission if you include the service in the pbxs you are selling. Well, not free at all. Note: Being an ActiveX component, please use internet explorer. And not having too much to do with open either, I guess. I would have called you to personally tell you that your mail was misplaced (there is some kind of asterisk-biz
[asterisk-users] Newcomer Question
Hallo Martin and Group! Thank you very much for your perfect introduction into asterisk. I managed to * get asterisk server running * configuring the internal numbers * registering to 2 sip gateways * outbound phoning to sipgate works perfect * outbound phoning to mujtelefon not yet tested The problem i am having now is, that i cant be reached by inbound phone calls from neither sipgate nor mujtelefon i used my mobile to call this numbers. sipgate tells me on the phone:Das Endgeraet ist fuer diesen Service nicht konfiguriert. bei mujtelefon kommt nur die mobilbox In asterisk cli i dont see anything about that. in the sipgate login page there is neither mentioned. If this is working, I intend to have nice music played for incoming calls until the phone call is accepted. What is very confusing for sipgate is, that my number(734365) is different from my user name(1734365). Can anybopdy check, if all settings are ok according to that ? Please find below my sip.conf, only the passwords are scrambled. If you directly reply to me, also reply to [EMAIL PROTECTED] because i am afraid missing your answer in the much traffic in that mailing list. Thank you very much! [general] bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm musicclass=default language=de dtmfmode=rfc2833 sipdebug=no register = 1734365:[EMAIL PROTECTED]:5060/00437201734365 register = 272048160:[EMAIL PROTECTED]:5060/00420272048160 [sipgateat] host=sipgate.at secret=NMMTNMKP username=1734365 fromuser=1734365 fromdomain=sipgate.at srvlookup=yes context=sipgateat-in canreinvite=no nat=no type=friend qualify=yes
Re: [asterisk-users] IAX Java Softphone?
Dean Collins a écrit : As far as I know Jiaxclient is dead - the developer hasn't touched it in at least 18 months. Correct, but this is free software, anybody with the skills can revive it :) Regards, -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO
Speaking for the Asterisk community as a whole, we demand that it be free forever. Please honor your statement below, We posted in this forum because it is a contribution to the asterisk community, and because it is free for a month, and maybe even longer if the community so demands it. C. Savinovich wrote: Please don't change the title of my post. It is disrespectful. One thing is to give your opinion about its content, and another to be self appointed editor of this forum. If you agree or disagree with it fine, but let others decide. They know spam when they see it. Thank you. C. Savinovich VideoReps.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Thursday, September 20, 2007 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO Am Donnerstag, den 20.09.2007, 11:30 -0700 schrieb C. Savinovich: Dear All: Just as the name suggests, and evolving from regular Click-to-Call, Click-to-Call WITH VIDEO provides web sites with the ability to engage their visitors with a live video agent (plus the phone call). All with just a click of a button placed on the customer's web site. Please visit us at www.videoreps.net. When I read this, I thougt: Wow, here comes a nice, free, open, interesting software. Best of all, it is free for now!! (video only)... Here is the deal: Free for one month, no commitments. Try, test it, call me. After a month you can decide if you want to keep any of our plans: Video-Call starts at $12.95 per month, and click-to-call with video at $29.95. And yes, there is 20% monthly commission if you include the service in the pbxs you are selling. Well, not free at all. Note: Being an ActiveX component, please use internet explorer. And not having too much to do with open either, I guess. I would have called you to personally tell you that your mail was misplaced (there is some kind of asterisk-biz list, and I do not read it for a purpose), but I do not use the integrated exploder for lack of the necessary obfuscation system on my work machine. Please do not send commercials, ads and product information to this list. It might very well be considered SPAM. Just and only because _some_ readers might be interested there is no legitimation for sending it (else every pen15-en1argm3nt _might_ trigger interest at some readers). Thanks Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [PHISH] Re: ***SPAM*** Announcing: Click-to-Call with VIDEO
Please honor your statement below, We posted in this forum because it is a contribution to the asterisk community, and because it is free for a month, and maybe even longer if the community so demands it. I take your request as an official demand from the community to provide the service for free for a longer time. We are happy to work with you guys. Herewith, for anyone who signs up until Sunday September 23, will get 3 months free. I will personally go through the database and extend everyone who has signed with the committed time. We sincerely think we were making our 2 cents contribution to the asterisk community by announcing an innovative concept on this forum. We apologize if we have inconvenienced anyone but are nevertheless glad to be of any help. Don't want to abuse our welcome here anymore, so we will be happy to respond to personal inquiries. Best Regards C. Savinovich -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, September 20, 2007 12:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [PHISH] Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO Speaking for the Asterisk community as a whole, we demand that it be free forever. Please honor your statement below, We posted in this forum because it is a contribution to the asterisk community, and because it is free for a month, and maybe even longer if the community so demands it. C. Savinovich wrote: Please don't change the title of my post. It is disrespectful. One thing is to give your opinion about its content, and another to be self appointed editor of this forum. If you agree or disagree with it fine, but let others decide. They know spam when they see it. Thank you. C. Savinovich VideoReps.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Thursday, September 20, 2007 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO Am Donnerstag, den 20.09.2007, 11:30 -0700 schrieb C. Savinovich: Dear All: Just as the name suggests, and evolving from regular Click-to-Call, Click-to-Call WITH VIDEO provides web sites with the ability to engage their visitors with a live video agent (plus the phone call). All with just a click of a button placed on the customer's web site. Please visit us at www.videoreps.net. When I read this, I thougt: Wow, here comes a nice, free, open, interesting software. Best of all, it is free for now!! (video only)... Here is the deal: Free for one month, no commitments. Try, test it, call me. After a month you can decide if you want to keep any of our plans: Video-Call starts at $12.95 per month, and click-to-call with video at $29.95. And yes, there is 20% monthly commission if you include the service in the pbxs you are selling. Well, not free at all. Note: Being an ActiveX component, please use internet explorer. And not having too much to do with open either, I guess. I would have called you to personally tell you that your mail was misplaced (there is some kind of asterisk-biz list, and I do not read it for a purpose), but I do not use the integrated exploder for lack of the necessary obfuscation system on my work machine. Please do not send commercials, ads and product information to this list. It might very well be considered SPAM. Just and only because _some_ readers might be interested there is no legitimation for sending it (else every pen15-en1argm3nt _might_ trigger interest at some readers). Thanks Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Paging MEETME_RECORDINGFILE Variable
I am having a weird issue with setting the recording file for the
Page app. Here is some quick background info
I have a macro that pages all my phones:
[macro-pageall]
; Context for paging all devices.
; This will search the sip table in the realtime database
; for all phones that start with a number. That number is
; passed to this macro as ${ARG1}.
;
; ARG1 = The first digit of the phones to be paged (6=US
Campus, 4=MS, 2=LS)
; ARG2 = Device for the PA system. If the user selected to
; page the PA system. That will be included.
;
exten = s,1,MYSQL(Connect connid ${realdb_host} ${realdb_user} $
{realdb_pass} ${realdb_db})
exten = s,2,MYSQL(Query resultid ${connid} SELECT\ name\ FROM\ sip\
WHERE\ name\ LIKE\ '${ARG1}%')
exten = s,3,MYSQL(Fetch fetchid ${resultid} number)
exten = s,4,GoToIf($[${fetchid} = 1]?5:7)
exten = s,5,Set(pagedevice=${pagedevice}SIP/${number})
exten = s,6,GoToIf($[${fetchid} = 1]?3:7)
exten = s,7,Set(pagedevice=${pagedevice:1})
exten = s,8,MYSQL(Clear ${resultid})
exten = s,9,MYSQL(Disconnect ${connid})
exten = s,10,GoToIf($[${ARG2} != ]?11:12)
exten = s,11,Set(pagedevice=${pagedevice}${ARG2})
;Add Call Info for GrandStream Phone on the PA system
exten = s,12,SIPAddHeader(Call-Info:answer-after=0)
;Add Alert-Info for all Polycom Phones
exten = s,13,SIPAddHeader(Alert-Info: Ring Answer)
exten = s,14,Set(MEETME_RECORDINGFILE=custom/paging/campuslastpage_$
{RAND(1|100)})
exten = s,15,NoOp(${MEETME_RECORDINGFILE})
exten = s,16,Set(CALLERID(all)=System Page 1010)
exten = s,17,Page(${pagedevice},r)
exten = h,1,System(/var/lib/asterisk/scripts/mail_lastpage ${ARG1} $
{MEETME_RECORDINGFILE})
exten = h,2,Hangup()
I call the macro with:
;Page All Phones including the PA system.
exten = 1010,1,Authenticate(12345)
exten = 1010,2,Macro(pageall,2,SIP/ls-pa)
Basically the macro goes through my sip realtime database and finds
all the phones that begin with the number 2 (my lower school
campus). The generates a variable named pagedevice that looks like
this:
SIP/2101SIP/2102SIP/2103
This part works great.
The issue I am having is setting the MEETME_RECORDINGFILE. It should
be set to an audio file in the custom sounds directory with a random
number at the end. I then use a hangup (h) extension to execute a
script (at bottom of email) to email the audio file to a conference
area in our email system (FirstClass).
What is weird is after I restart the asterisk process, this works
fine for about a week. It does exactly as it is supposed to, creates
the audio file with a random number, then the email script delivers
it. After a week or so Asterisk will stop setting the variable
MEETME_RECORDINGFILE and start placing the recordings in the sounds
directory named meetme-conf-rec.##.wav. Which is the default is
MEETME_RECORDINGFILE is not set.
Anyone seen this issue before?
Thanks!
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
#!/bin/bash
#Set some variables
USFACULTY=[EMAIL PROTECTED]
LSFACULTY=[EMAIL PROTECTED]
USFACULTY=[EMAIL PROTECTED]
MONTH=`date +%B`
DAY=`date +%d`
YEAR=`date +%Y`
HOUR=`date +%I`
MINUTE=`date +%M`
ZONE=`date +%Z`
AMPM=`date +%P`
PGSOUNDDIR=/var/lib/asterisk/sounds/
LOGFILE=/var/log/mail_lastpage.log
#Write Log
echo `date` Running script for campus $1 with file $2 $LOGFILE
#Let give asterisk time to finish creating the recordng file. Just
in Case.
sleep 10
#
#Create a temp file with our message body
#
echo Repeat Last Page /tmp/repeatpage_$1
echo /tmp/repeatpage_$1
echo The attached WAV file is a copy of the last broadcast over the
phone system. /tmp/repeatpage_$1
echo /tmp/repeatpage_$1
echo The page was broadcasted $MONTH $DAY, $YEAR at $HOUR:$MINUTE
$AMPM. You may play this file back if you missed the page. /tmp/
repeatpage_$1
echo /tmp/repeatpage_$1
echo /tmp/repeatpage_$1
echo If you wish to mark this email as read (Remove Red Flag)
without opening the email, you may right-click (or control-click for
Mac) and left-click Mark
as Read before opening the email. /tmp/repeatpage_$1
#
#Send the email with the recorded Page attached
#
# Was it Upper School?
if [ $1 -eq 6 ]
then
cat /tmp/repeatpage_$1 | mutt -a $PGSOUNDDIR$2.wav -
s Recording of Last Page for Upper School $USFACULTY
fi
# Was it Middle School?
if [ $1 -eq 4 ]
then
cat /tmp/repeatpage_$1 | mutt -a $PGSOUNDDIR$2.wav -
s Recording of Last Page for Middle School $MSFACULTY
fi
# How about Lower?
if [ $1 -eq 2 ]
then
cat /tmp/repeatpage_$1 | mutt -a $PGSOUNDDIR$2.wav -
s Recording of Last Page for Lower School $LSFACULTY
fi
rm -rf /tmp/repeatpage_$1
rm -f $PGSOUNDDIR$2.wav
exit
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To
Re: [asterisk-users] Polycom 330 + Asterisk, phone locks up. * key will do it
[EMAIL PROTECTED] wrote: Hello all, We have an Asterisk server that has worked without issue for a while. Before, only Sipura and Polycom 500 series phones were used. Recently, we've added a few POE switches and 20 or so Polycom 330's. The 330's seem to lock up often. One easy way to do this is by hitting the * key. I looked thru the XML files for dialplan and what not, nothing stood out. I've tried the latest everyone-can-download (bootrom 3.2.3.0021 and firmware 2.1.2.0048). Thoughts? The 500 series phones (501) have no issues. Without a doubt the most common problem with deploying newer Polycom handsets is the presence of old configurations instructions in the XML provisioning files. In most cases, people plug in a 650, provision it just like they did with all their 501s, and the reliability is just, well, .. terrible! The degree of pain depends on how (and how much) you have tweaked the configs over the years. The fix can be a little painful, but is worth doing. Upgrade to the latest firmware and XML config templates from Polycom (assuming you purchased from an official polycom reseller), and port your local changes as overrides from the new default file. Config files are backward compatible, but not forward compatible. What worked with your 501 will not make your 430, 650, 330 or 320 happy. -Darren -- Darren Nickerson Senior Sales Support Engineer Telephony Depot www.telephonydepot.com +1.215.825.8710 ext 8106 (office) +1.215.243.8335 (fax) ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GROUP() issues for me
I've got a macro that tries to find the first available SIP trunk to send
outgoing calls on. It tracks the usage of the lines (since each trunk has a
call-limit of 2) by using GROUP(). My problem is that once a call switched
to ANSWER state, ``group show channels`` stops listing it and then my Macro
starts screwing up because it's sending calls to a line that sometimes is
full even though GROUP() shows it as being less than 2. I'm tempted to send
this to the Asterisk Dev team just because I believe it's an issue of the
GROUP information being released when Asterisk consolidates the channels
(removes all the MASQ channels) once the call is answered. But maybe it's
something else so I'll ask here first.
The dialplan setup:
exten = 555,1,Dial(Local/1234567890)
exten = _NXXNXX,1,Macro(which-line,${EXTEN})
[macro-which-line]
exten = s,1,set(GROUP()=${DIALSTRING})
exten = s,n,Dial(${DIALSTRING}/1${ARG1})
Things are a bit more complex, but it's all just logic. The extensions
above should give a decent representation of what's going on. I think each
time you switch extensions, Asterisk creates a MASQ channel and that's
what's causing the issue since the GROUP() is set only at the end, inside
the macro. Are there any EVENTS for unlocking of GROUPs? Anything I can do
to better show where this is happening?
I'd love some help if anyone has a guess.
--
/Nick
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.24 simultaneous call limits.
Hi everyone, I am running into wall today with simultaneous call limits. I have two Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to create a lot of sip calls from one machine to the other by issuing AMI Originate commands to one machine. The machine that makes calls plays a message (demo-intruct) upon the other machine answer. The machine receives the calls just waits for 40 seconds then hangs up. Throught the manager connection, I was creating 10 calls per-second. I also have sip phone registered with the calling machine. At around 150 to 200 calls. When I call the machine that's making all the calls, most of the calls couldn't go through. For the ones that went through, most of them will drop off within seconds of the call. But here is catch. When I run 'top', the cpu is idling 97%. My question is. Is there a limit on the number of simultaneous calls Asterisk can handle? I know I have very fast systems. Shouldn't they be able to handle that many calls? What is your take? Thnx ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GROUP() issues for me
try
Nicholas Blasgen wrote:
I've got a macro that tries to find the first available SIP trunk to send
outgoing calls on. It tracks the usage of the lines (since each trunk has a
call-limit of 2) by using GROUP(). My problem is that once a call switched
to ANSWER state, ``group show channels`` stops listing it and then my Macro
starts screwing up because it's sending calls to a line that sometimes is
full even though GROUP() shows it as being less than 2. I'm tempted to send
this to the Asterisk Dev team just because I believe it's an issue of the
GROUP information being released when Asterisk consolidates the channels
(removes all the MASQ channels) once the call is answered. But maybe it's
something else so I'll ask here first.
The dialplan setup:
exten = 555,1,Dial(Local/1234567890)
exten = 555,1,Dial(Local/1234567890/n)
note the /n
exten = _NXXNXX,1,Macro(which-line,${EXTEN})
[macro-which-line]
exten = s,1,set(GROUP()=${DIALSTRING})
exten = s,n,Dial(${DIALSTRING}/1${ARG1})
Things are a bit more complex, but it's all just logic. The extensions
above should give a decent representation of what's going on. I think each
time you switch extensions, Asterisk creates a MASQ channel and that's
what's causing the issue since the GROUP() is set only at the end, inside
the macro. Are there any EVENTS for unlocking of GROUPs? Anything I can do
to better show where this is happening?
I'd love some help if anyone has a guess.
that's my guess
Julian
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GROUP() issues for me
exten = 555,1,Dial(Local/1234567890/n) note the /n I'm going to try this in a bit (can't hurt anything, might as well), but I'd like to understand you're reasoning. You're dialing an extra extension? I'm also going to be trying this with Asterisk 1.6 TRUNK to see if it's even a current issues in the development branch but I wont have a chance untill tomorrow sometime. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.24 simultaneous call limits.
Just thinking about it quickly, it's always possible it has nothing to do with Asterisk. There are many instances where I run into issues with a poorly configured servers when they have even a little bump in HTTP traffic. This was years ago though, and it was an issue to do with a web server and not Asterisk, but look into your kernel's configuration. Sometimes the kernel's settings are setup for a normal USER and not designed to handle the memory allocation a server demands. The fix for me back then was something to do with the MAXIMUM PAGE REQUESTS or SIZE maybe. Basicly the kernel couldn't keep track of all the HTTP processes. Now that I'm reading this over I doubt it's your problem because Asterisk doesn't fork. But while we're at it, tell me a bit more about your system. What operating system (and version)? The problem could also be with your method of load generation, but I wouldn't know that since I've never tried load testing a system. Lastly, I know FreeBSD started incorporating a basic DDoS protection a few years back and maybe that's also in some of these newer Linux distros. They would detect a flood and start to limit the bandwidth. These are just ideas, I don't really like any of them. Sometimes the kernel will report issues to SYSLOGD. Might want to check your error and message logs. cat /proc/meminfo On a Linux box will give you memory limits and how close you are to them. They're not exactly what I was looking for, but maybe that will help. All TCP connections require the Kernel to page the information but I can't seem to find out how to access that limit if any. On 9/20/07, Wai Wu [EMAIL PROTECTED] wrote: Hi everyone, I am running into wall today with simultaneous call limits. I have two Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to create a lot of sip calls from one machine to the other by issuing AMI Originate commands to one machine. The machine that makes calls plays a message (demo-intruct) upon the other machine answer. The machine receives the calls just waits for 40 seconds then hangs up. Throught the manager connection, I was creating 10 calls per-second. I also have sip phone registered with the calling machine. At around 150 to 200 calls. When I call the machine that's making all the calls, most of the calls couldn't go through. For the ones that went through, most of them will drop off within seconds of the call. But here is catch. When I run 'top', the cpu is idling 97%. My question is. Is there a limit on the number of simultaneous calls Asterisk can handle? I know I have very fast systems. Shouldn't they be able to handle that many calls? What is your take? Thnx ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /Nick ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Question
Hi Jeremy,
A few thoughts that come to mind. We have a queue that is open between
certain hours. I have a few checks in place before a caller enters,
first it checks to see if there it is within the time window, then
checks to see if there are any agents log into queue, if any fail they
get our closed message. Sounds like you are trying to do something similar.
Not sure what you have for extension numbers numbers, but you will get
the idea.
Your first friend:
GotoIfTime(time range|days of week|days of
month|months?[[context|]extension|]pri)
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime
I don't know how your dial plan is structured. My guess is the after
hours operation is in a separate part of the code from the other. Since
we are just looking at after hours, I would use the reverse on your
time. Because the command jumps when the statement is true. I do not
know what will happen if you say go from 17:00-8:00, but you can try it.
Example:
exten = 800,1,GotoIfTime(8:00-17:00|mon-fri|*|*?NormalOp,900,1) ; Since
this will fail if it is 9pm, it moves on to the next priority in this
exten.
[NormalOp]
exten = 900,1,blah
Next, is your other test. Use the queue agent count function
QUEUEAGENTCOUT(queuename)
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+queueagentcount
If the number is greater then 0, then you move them into the queue, if
not, whatever you want.
Finally, in terms of your other questions about logging the agents in.
You could do the database way. You also could create a log in extension
where you can take their cell number ( caller id) and use the
application AddQueueMember(queuename[|interface][|penalty])
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AddQueueMember
So you should be able to do something like
AddQueueMember(queueName|ZAP/${CALLID(num)})
Anyway hope that helps.
Kevin
Jeremy Mann wrote:
I’m curious if anyone has implemented the following:
Need to setup an on-call queue, that activates after 5PM and
de-activates at 8AM, also that activates/deactivates on demand(I’m
thinking a feature code here). The “agents” need to log in via cell
phones, and when calls come in from outside to the asterisk system,
it’ll need to call the cell phone agents that are active.
I’m thinking that it’s a simple SQL query, to update the agents status
and number, and that asterisk will do a lookup and append that to the
ZAP channel to dial, but interested in any logic someone might be able
to come up with for the dialplan.
This e-mail, facsimile, or letter and any files or attachments
transmitted with it contains information that is confidential and
privileged. This information is intended only for the use of the
individual(s) and entity(ies) to whom it is addressed. If you are the
intended recipient, further disclosures are prohibited without proper
authorization. If you are not the intended recipient, any disclosure,
copying, printing, or use of this information is strictly prohibited
and possibly a violation of federal or state law and regulations. If
you have received this information in error, please notify Texas
Health Management Group immediately at 1-817-310-4999. Texas Health
Management Group, its subsidiaries, and affiliates hereby claim all
applicable privileges related to this information.
--
This message has been scanned for viruses and
dangerous content by *MailScanner* http://www.mailscanner.info/, and is
believed to be clean.
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.24 simultaneous call limits.
Interesting. I am using PCLinuxOS(Mandrak) in console mode. Here is my memory info and you can see that I still have a lot of memory while asterisk is running [EMAIL PROTECTED] ~]# cat /proc/meminfo MemTotal: 2076000 kB MemFree: 1855636 kB Buffers: 17224 kB Cached: 115916 kB SwapCached: 0 kB Active: 124100 kB Inactive:73468 kB HighTotal: 1179264 kB HighFree: 992808 kB LowTotal: 896736 kB LowFree:862828 kB SwapTotal: 1365484 kB SwapFree: 1365484 kB Dirty: 216 kB Writeback: 0 kB AnonPages: 64524 kB Mapped: 43912 kB Slab:13168 kB PageTables: 1344 kB NFS_Unstable:0 kB Bounce: 0 kB CommitLimit: 2403484 kB Committed_AS: 142512 kB VmallocTotal: 114680 kB VmallocUsed: 8484 kB VmallocChunk: 104492 kB [EMAIL PROTECTED] ~]# You mentioned DDoS projection. How can I find out if my distro has it built in? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicholas Blasgen Sent: Thursday, September 20, 2007 8:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.2.24 simultaneous call limits. Just thinking about it quickly, it's always possible it has nothing to do with Asterisk. There are many instances where I run into issues with a poorly configured servers when they have even a little bump in HTTP traffic. This was years ago though, and it was an issue to do with a web server and not Asterisk, but look into your kernel's configuration. Sometimes the kernel's settings are setup for a normal USER and not designed to handle the memory allocation a server demands. The fix for me back then was something to do with the MAXIMUM PAGE REQUESTS or SIZE maybe. Basicly the kernel couldn't keep track of all the HTTP processes. Now that I'm reading this over I doubt it's your problem because Asterisk doesn't fork. But while we're at it, tell me a bit more about your system. What operating system (and version)? The problem could also be with your method of load generation, but I wouldn't know that since I've never tried load testing a system. Lastly, I know FreeBSD started incorporating a basic DDoS protection a few years back and maybe that's also in some of these newer Linux distros. They would detect a flood and start to limit the bandwidth. These are just ideas, I don't really like any of them. Sometimes the kernel will report issues to SYSLOGD. Might want to check your error and message logs. cat /proc/meminfo On a Linux box will give you memory limits and how close you are to them. They're not exactly what I was looking for, but maybe that will help. All TCP connections require the Kernel to page the information but I can't seem to find out how to access that limit if any. On 9/20/07, Wai Wu [EMAIL PROTECTED] wrote: Hi everyone, I am running into wall today with simultaneous call limits. I have two Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to create a lot of sip calls from one machine to the other by issuing AMI Originate commands to one machine. The machine that makes calls plays a message (demo-intruct) upon the other machine answer. The machine receives the calls just waits for 40 seconds then hangs up. Throught the manager connection, I was creating 10 calls per-second. I also have sip phone registered with the calling machine. At around 150 to 200 calls. When I call the machine that's making all the calls, most of the calls couldn't go through. For the ones that went through, most of them will drop off within seconds of the call. But here is catch. When I run 'top', the cpu is idling 97%. My question is. Is there a limit on the number of simultaneous calls Asterisk can handle? I know I have very fast systems. Shouldn't they be able to handle that many calls? What is your take? Thnx ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- /Nick ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GROUP() issues for me
Nicholas Blasgen wrote: exten = 555,1,Dial(Local/1234567890/n) note the /n I'm going to try this in a bit (can't hurt anything, might as well), but I'd like to understand you're reasoning. You're dialing an extra extension? I'm also going to be trying this with Asterisk 1.6 TRUNK to see if it's even a current issues in the development branch but I wont have a chance untill tomorrow sometime. read localchannel.txt or whatever obvious file in your version of Asterisk. /path/to/src/asterisk/docs/ Also, you need an @context, for example: exten = 555,1,Dial(Local/[EMAIL PROTECTED]/n) Where corporate is whatever context the extension 1234567890 is located in. Maybe not needing a context is new in 1.4, but the doc file would tell you. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Samsung Sprint CDMAoIP
On 9/20/07, Jason Parker [EMAIL PROTECTED] wrote: C F wrote: AFAIK, the calls are free when you use it thru that device. Sprint however charges $15 a month per phone or $30 for family plan. While I agree that sprint should pay me for this, it's not as bad. T-mobile on the other hand, does the same thing with wifi enabled phones, it doesn't cost extra, and is completely free. If you're referring to T-Mobile's [EMAIL PROTECTED] service, it's actually $20 per month, per line on the account (unless it's changed very recently). I don't know about that, could be you are right. As far as how it works on T-Mobile, I recently had some questions answered by them about that.. They use UMA over wifi, and it does automatic switching between the wifi and the gsm towers (ie; your call stays up). The same goes for Sprint. Quote from the tech I talked to: [EMAIL PROTECTED] does not use a VoIP protocol, as the voice data is transferred from the Internet directly to our UMA Gateway and then through our regular Mobile Switching Centers. I know it's a quote from the tech, but isn't it voice packets that travels over the Internet (a packet switched network) instead of over GSM (TDM switched network) which makes that statement incorrect? It doesn't matter what the higher level protocol is, it's still VoIP. Pretty interesting stuff. -- Jason Parker Digium ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialing an external number and then passing it to an extension...
I am in need of some guidance regarding the following problem:
I need to dial an external number from a list(PSTN)
I need to check if the number is busy, no answer or fail
If any of the above are met then I try another number from a list
If none of the above happen then I first need to determine if the line
answering is a fax machine or an answering machine
If fax or answering machine then hangup and try next number
If human then connect to an internal extension
An outbound callcenter suite is overkill since we only need two or three
calls at a time. Can something like this be done using the Originate command
on AMI? The main problem I have is that if I dial an external call and it
fails for some reason how do I know? Is there something like ${DIALSTATUS}
that can give me the result of that part of the call?
We plan to have a web interface that will fire the call when you click a
button. That will fire an event that connects to the manager interface and
uses originate to dial the external call and then dial the internal extension
if all conditions are met. The numbers will be in a database.
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Samsung Sprint CDMAoIP
Snip headers On 9/20/07, Jason Parker [EMAIL PROTECTED] wrote: C F wrote: AFAIK, the calls are free when you use it thru that device. Sprint however charges $15 a month per phone or $30 for family plan. While I agree that sprint should pay me for this, it's not as bad. T-mobile on the other hand, does the same thing with wifi enabled phones, it doesn't cost extra, and is completely free. If you're referring to T-Mobile's [EMAIL PROTECTED] service, it's actually $20 per month, per line on the account (unless it's changed very recently). I don't know about that, could be you are right. As far as how it works on T-Mobile, I recently had some questions answered by them about that.. They use UMA over wifi, and it does automatic switching between the wifi and the gsm towers (ie; your call stays up). The same goes for Sprint. Quote from the tech I talked to: [EMAIL PROTECTED] does not use a VoIP protocol, as the voice data is transferred from the Internet directly to our UMA Gateway and then through our regular Mobile Switching Centers. I know it's a quote from the tech, but isn't it voice packets that travels over the Internet (a packet switched network) instead of over GSM (TDM switched network) which makes that statement incorrect? It doesn't matter what the higher level protocol is, it's still VoIP. Your right it is STILL VoIP by definition but its not... From: http://www.newstep.com/our%20market/technologies.asp Gateway-based Solutions By placing special gateways at the edge of a GSM network, Unlicensed Mobile Access (UMA) allows users with dual-mode handsets to access mobile phone services via both cellular and Wi-Fi links. In cellular mode, voice traffic travels over standard GSM radio waves. In Wi-Fi mode, an IP tunnel carries GSM traffic across the enterprise network and/or the Internet to a UMA gateway. The gateway looks like a base station controller (BSC) to the cellular network, so when a handset moves between cellular and Wi-Fi coverage, the network handles it as an ordinary BSC-to-BSC handoff. MSC emulation-also known as IP VLR-is similar to UMA, except that the gateway mimics a mobile switching center (MSC) and a visitor location register (VLR) instead of a BSC. Intimately tied to cellular technology and dual-mode handsets, gateway-based solutions provide access only to mobile network services and can be deployed only by facilities-based mobile network operators. Moreover, gateway-based solutions cannot leverage the full capabilities of IP and VoIP because all voice traffic remains in TDM format. Service providers, therefore, view gateway-based solutions as an inadequate response to the FMC opportunity. They are turning instead to server-based technology, a more generalized approach that spans all types of networks: fixed and mobile, IP and TDM, business and residential. Pretty interesting stuff. -- Jason Parker Digium ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
