[asterisk-users] g729 open source codec and sample size
Greetings. I'm new to the asterisk voip world and I'm currently trying out trixbox 2.6.0.7 on a p4 1.8 GHz box. I've downloaded and used the open source g729 codec from site http://asterisk.hosting.lv/ and is working fine. question here is that this codec sends out a packet every 20ms. Though the speech quality is very good, I also like to try out 30ms sampling size to bring down the overhead payload and reduce bandwidth usage. I've searched for it for a couple days with no indication of how to do it. is it possible to change it. do i have to compile my own codec module.. or some patch to asterisk code?? Please suggest. Thanks a lot. Manoj -- ___ -- 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] odd error compiling zaptel-1.4.10 - XPP
http://bugs.digium.com/12426 There's also a fix there that I don't fully understand (and I'm not sure that that fix does not cause damage, so don't just apply it). I am installing Zaptel 1.4.10.1 and I encountered the same problem. As Jerry said, compile problem goes away if I uncheck xpp in make menuselect. By unchecking xpp, as long as I don't use Xorcom Astribank for USB channel bank, then there won't be a problem. Is this correct? ___ -- 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] odd error compiling zaptel-1.4.10 - XPP
On Tue, Jun 10, 2008 at 05:40:10PM +1000, Lee, John (Sydney) wrote: http://bugs.digium.com/12426 There's also a fix there that I don't fully understand (and I'm not sure that that fix does not cause damage, so don't just apply it). I am installing Zaptel 1.4.10.1 and I encountered the same problem. As Jerry said, compile problem goes away if I uncheck xpp in make menuselect. By unchecking xpp, as long as I don't use Xorcom Astribank for USB channel bank, then there won't be a problem. Is this correct? If you look at that bug report, you'll see that the error message there happens to be completely harmless to anybody without an Astribank (and not even for all of those with Astribanks). So you can just ignore it. While in menuselect, you can probably uncheck most of the modules to save you compilation time. e.g: to get rid of the need to re-build voicebus.o each time you run 'make' or 'make install'. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Fax on FXS
On Mon, Jun 09, 2008 at 06:53:34PM -0400, Matt Watson wrote: On June 9, 2008 01:34:31 pm Eric ManxPower Wieling wrote: You should not expect FaxOverVoiceOverIPOverInternet to work well. If you stick to ulaw codec for the entire call, it might work well enough for your use, but it might not. Just as an FYI - you have too many Over's in your description FaxOverVoiceOverIP would make sense, but seeing as how IP is short for Internet Protocol, saying Internet Protocol Over Internet doesn;t make much sense... Unless you use an openvpn / ipsec tunnel :-) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] meetme recording with security?
Hi: I configured an asterisk server for conference call service but I have a problem now :Does asterisk have an option to secure and warranty meetme,in the other word,How can I play up users that their conference won't hear by us in spite of asterisk can record meetme ? I'd appreciate any help. ___ -- 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] Help-ASTERISK-MFCR2
Alvaro, we've already set debug level at 255 on unicall.conf and at logger.conf we've enabled full log notice,warning,error,debug,verbose). The log console output is: Here is the LOGS when I try do make calls, the call will not go to Asterisk Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 0001 [1/ 1/Idle /Idle ] Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Detected Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Making a new call with CRN 32769 Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1101 - [2/ 2/Idle /Idle ] Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Detected Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1001 [2/ 2/Seize ack /Seize ack] Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Far end disconnected(cause=Normal, unspecified cause [31]) - state 0x2 Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Far end disconnected Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2930 handle_uc_event: CRN 32769 - far disconnected cause=Normal, unspecified cause [31] Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(6) Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Drop call(cause=Normal Clearing [16]) Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call disconnected(cause=Normal, unspecified cause [31]) - state 0x800 Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Drop call Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(7) Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Release call Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/1000/Clear fwd /Seize ack] Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Release guard expired Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Destroying call with CRN 32769 Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Release call -- Unicall/1 released Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Thanks ... Regards, Mariano - Original Message - From: Alvaro Parres To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, June 09, 2008 7:04 PM Subject: Re: [asterisk-users] Help-ASTERISK-MFCR2 Mariano: Could you send us please the log files, and the console output... so we can help you. On Mon, Jun 9, 2008 at 8:01 AM, Mariano Borgognone [EMAIL PROTECTED] wrote: Moises, we've already set debug level at 255 on unicall.conf and at logger.conf we've enabled full log (notice,warning,error,debug,verbose). Has anyone experienced with a Siemens EWSD switch? Anyone knows about to change R2 timers at unicall.conf ? Please any comment is welcome, thank you.. Mariano.- - Original Message - From: Moises Silva [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, June 07, 2008 1:27 PM Subject: Re: [asterisk-users] Help-ASTERISK-MFCR2 You need to enable loglevel=255 in unicall.conf and enable all the levels of logging in logger.conf, otherwise the logs you post don't say much. Moisés Silva On Fri, Jun 6, 2008 at 2:58 PM, Mariano Borgognone [EMAIL PROTECTED] wrote: Dears, I have problem ASTERISK with PSTN SIEMENS EWSD (MFC R2), I don´t receive call for PSTN, i don´t understand why. please i need your help # MFC/R2 normalmente no usa CRC4 span=1,1,0,cas,hdb3 cas=1-15:1101 dchan=16 cas=17-31:1101 loadzone=us defaultzone=us [channels] usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,10 protocolend=cpe group = 1 context= e1-incoming channel = 1-15 channel = 17-31 ;skip time slot 16 Here is the LOGS when I try do make calls Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 -
Re: [asterisk-users] redfone fonebridge2
Bill Michaelson wrote: I'm looking for reports of recent experience with redfone fonebridge2 (with echo can) TDMoE gizmos. Anybody? Good? Bad? We use it and it works without any problems. Tech support was helpful, documentation was not. Thumbs up. ___ -- 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] Fax on FXS
Ah, you got me there! Could start throwing in a lot of Over's going down that road :) -- Matt http://www.mattgwatson.ca -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Tuesday, June 10, 2008 4:10 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Fax on FXS On Mon, Jun 09, 2008 at 06:53:34PM -0400, Matt Watson wrote: On June 9, 2008 01:34:31 pm Eric ManxPower Wieling wrote: You should not expect FaxOverVoiceOverIPOverInternet to work well. If you stick to ulaw codec for the entire call, it might work well enough for your use, but it might not. Just as an FYI - you have too many Over's in your description FaxOverVoiceOverIP would make sense, but seeing as how IP is short for Internet Protocol, saying Internet Protocol Over Internet doesn;t make much sense... Unless you use an openvpn / ipsec tunnel :-) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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 ___ -- 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] meetme recording with security?
There's so such thing as privacy on a phone call, at least not in the United States since warrantless wiretapping. Then expecting security on a conference call, which by definition is open to many parties, is silly. Depending on your state (in the US), you may need to disclose when and if you do record calls, but nobody can reasonably expect security if they're using a public phone system in the US. You can tell the users of a service that calls will not be recorded by the service provider, but that's the only claim you can make in good faith. And while you're speaking of MeetMe's ability to record or snoop, don't forget ChanSpy(), and other features depending on if we're talking SIP or Zaptel, etc. -Dave On Tue, Jun 10, 2008 at 6:16 AM, fateme fatah [EMAIL PROTECTED] wrote: Hi: I configured an asterisk server for conference call service but I have a problem now :Does asterisk have an option to secure and warranty meetme,in the other word,How can I play up users that their conference won't hear by us in spite of asterisk can record meetme ? I'd appreciate any help. ___ -- 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 ___ -- 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 : using setvar with IP Realtime and variable inheritance
callerid_internal=test 710;did=551234
Again, this works fine. The problem is when I forward my calls to another
outside line (using Polyocm phones), and need to know the ${did} value at
that point. It's empty.
The other answer looks pretty good. If that doesn't work, do a sip
debug on your console, and see if the values you want are in the
traffic at all. You can set up some more complicated rules to parse
your SIP headers if the values are just in a different field.
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Re: [asterisk-users] Asterisk : using setvar with IP Realtime and variable inheritance
Mike wrote:
If I hardcode this value in my dialplan using two underscores before it (i.e
Setvar(__did=551234) ) this works. But I can't hardcode it, I need to
fetch it from the table.
Have you tried:
Set(__did=${did})
That might work.
--
Leif Madsen
http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk
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Re: [asterisk-users] Polycom SIP and DHCP problem
On Jun 9, 2008, at 2:29 PM, Lyndon Griffin wrote: Apologies - I know this isn't either Polycom or ISC support, but if anyone would have an answer to my problem, I'm certain they would be on this list. I'm experiencing odd behavior with Polycom handsets obtaining DHCP addresses. It always worked fine for me up until a few months ago. Unfortunately, I can't narrow down when it stopped working, or why. All my Polycoms now appear to ignore my DHCP server entirely, according to the following pattern: Polycom - DHCPDISCOVER Server - DHCPOFFER on the correct network Polycom - DHCPREQUEST on the wrong network Server - DHCPNAK Polycom - Rinse, repeat ad infinitum Had the same issue a year or so ago - it related to a code version on the Polycoms. We wiped the flash and let them reload software I think. dont think we changed code but that took care of the issue. This was on one of our IP430 installs, never had it happen with 6xx series - yet. ___ -- 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] Debugging SIP call hangup reasons
Hi, Is there any information that can be gathered from the logs about why a SIP call was dropped/terminated without either side hanging up? I've run asterisk pretty verbose and I guess I haven't seen anything that pops out at me yet. I'm trying to diagnose why some clients are getting dropped calls every so often. Thanks. -- James ___ -- 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] Camp / Callback feature in 1.4
Hello I'm looking for a way to do the following using my Asterisk system and Snom SIP phones... Scenario: Caller on Internal Phone 1 calls internal phone2. Phone 2 is busy (or more accurately goes straight to voicemail). Caller on internal phone 1 can press a button / dial a code (explained in next step) and hangup When phone 2 is free, phone 1 rings and on answer dials phone 2 I was sure this was called camping - but all the camping stuff I can find, refers to the caller having to hang on the phone and wait. Am I missing something? Anyone have a solution? Thanks in advance Phil ___ -- 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] Polycom SIP and DHCP problem
Jerry Jones wrote: Had the same issue a year or so ago - it related to a code version on the Polycoms. We wiped the flash and let them reload software I think. dont think we changed code but that took care of the issue. This was on one of our IP430 installs, never had it happen with 6xx series - yet. Unfortunately, I've tried that... I now have various BootROM versions and multiple SIP versions - 1.6.x, 2.2.x, 3.0.0, 3.0.2c, on IP/301 IP/501 and IP/330 phones, all experiencing the same trouble. I've also now tried different ISC DHCPd versions, on different platforms, and have also tried W2k3's DHCP service. I will believe it's a code problem - I see that the phones are picking up *some* of the attributes I pass in DHCP offers, like the domain name. Not only that, but I've sniffed a phone actually trying to ARP the address the server DHCPOFFERED to it, before it decides to use a 192.168.0.x address. Can you tell me exactly which versions you have working? ___ -- 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] Camp / Callback feature in 1.4
On Tue, Jun 10, 2008 at 5:34 PM, Phil Knighton [EMAIL PROTECTED] wrote:
Hello
I'm looking for a way to do the following using my Asterisk system and Snom
SIP phones...
Scenario:
Caller on Internal Phone 1 calls internal phone2. Phone 2 is busy (or more
accurately goes straight to voicemail).
Caller on internal phone 1 can press a button / dial a code (explained in
next step) and hangup
When phone 2 is free, phone 1 rings and on answer dials phone 2
I was sure this was called camping - but all the camping stuff I can find,
refers to the caller having to hang on the phone and wait. Am I missing
something?
Anyone have a solution?
Quick solution that comes into mind:
Set(exten_copy = ${EXTEN});
Dial(SIP/${EXTEN})
if (${DIALSTATUS}=BUSY) {
// prompt for camp
Set(DB(camp/${EXTEN}/call_to)=${CALLERID(num));
}
h = {
Set(call_to=${DB(camp/${exten_copy}/call_to)});
if (${call_to}!=) {
Set(DB(camp/${exten_copy}/call_to)=);
System(call_to ${exten_copy} ${call_to});
}
}
So, in case if phone2 is busy, store callerid of phone1 in database,
so when phone2 will hangup it will triger a script call_to which
however can originate call trough manager or call-file.
Of course you will need some additional handling in case if multiple
callers decide to camp, or diferent protocols are used, etc.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] g729 open source codec and sample size
Manoj_Rajkarnikar wrote: Greetings. I'm new to the asterisk voip world and I'm currently trying out trixbox 2.6.0.7 on a p4 1.8 GHz box. I've downloaded and used the open source g729 codec from site http://asterisk.hosting.lv/ and is working fine. question here is that this codec sends out a packet every 20ms. Though the speech quality is very good, I also like to try out 30ms sampling size to bring down the overhead payload and reduce bandwidth usage. I've searched for it for a couple days with no indication of how to do it. is it possible to change it. do i have to compile my own codec module.. or some patch to you need to use the following parameter in your sip definitions (not sure if Trixbox will take it though) disallow=all allow=g729:30;30 is the frame size in ms Andres http://www.neuroredes.com asterisk code?? Please suggest. Thanks a lot. Manoj ___ -- 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] Interoffice phone setup
Asterisk gets very upset if it can't lookup the host name associated with every IP on the system, normally it would use DNS to do this, but since your Internet connection was down it could not do that. You should look at /etc/hosts on the Asterisk machine and make sure that each IP address of the system is listed and a name associated with it. You may have the change the order of the items in /etc/nsswitch to make sure file is consulted before dns. Joseph L. Casale wrote: The exact question pose I must leave for others to answer. However, I recently completed a project that overcomes the situation you describe. I installed a cellular gateway giving me a wireless trunk. If I lose IP connectivity I can route calls out through my cell carrier. Works really well. Appreciate the quick response! What I am concerned about is that there are maybe two problems:) Is that behavior at least normal? I don't want to wait until start of business to find out connectivity is up but phones aren't. Just seems odd. Thanks! jlc ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] Fax on FXS
IP can be run over many things. Internet, Local LAN, Corporate WAN, VPN, etc Each of these things have different characteristics and so I add them to the list. FaxOverVoiceOverIPOverLAN is something that has a good chance of working, as a LAN tends to have little latency and little jitter, where FaxOverVoiceOverIPOverInternet has neither low latency nor low jitter. How would suggest I indicate global internet instead of IP on a local lan? Matt Watson wrote: Ah, you got me there! Could start throwing in a lot of Over's going down that road :) -- Matt http://www.mattgwatson.ca -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Tuesday, June 10, 2008 4:10 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Fax on FXS On Mon, Jun 09, 2008 at 06:53:34PM -0400, Matt Watson wrote: On June 9, 2008 01:34:31 pm Eric ManxPower Wieling wrote: You should not expect FaxOverVoiceOverIPOverInternet to work well. If you stick to ulaw codec for the entire call, it might work well enough for your use, but it might not. Just as an FYI - you have too many Over's in your description FaxOverVoiceOverIP would make sense, but seeing as how IP is short for Internet Protocol, saying Internet Protocol Over Internet doesn;t make much sense... Unless you use an openvpn / ipsec tunnel :-) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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 ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 open source codec and sample size
The G729 codec is neither open source, nor is it free, and the license/patent does not make an exception for educational use. The Intel LIBRARIES are free for educational/personal use, but the license for that software says that you still need a license from the G729 patent holder before use. I don't understand why people won't pay $10/channel for a fully licensed, legal, and Asterisk supported G729 codec. Manoj_Rajkarnikar wrote: Greetings. I'm new to the asterisk voip world and I'm currently trying out trixbox 2.6.0.7 on a p4 1.8 GHz box. I've downloaded and used the open source g729 codec from site http://asterisk.hosting.lv/ and is working fine. question here is that this codec sends out a packet every 20ms. Though the speech quality is very good, I also like to try out 30ms sampling size to bring down the overhead payload and reduce bandwidth usage. I've searched for it for a couple days with no indication of how to do it. is it possible to change it. do i have to compile my own codec module.. or some patch to asterisk code?? Please suggest. Thanks a lot. Manoj -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] Camp / Callback feature in 1.4
snip
Set(exten_copy = ${EXTEN});
Dial(SIP/${EXTEN})
if (${DIALSTATUS}=BUSY) {
// prompt for camp
Set(DB(camp/${EXTEN}/call_to)=${CALLERID(num));
}
h = {
Set(call_to=${DB(camp/${exten_copy}/call_to)});
if (${call_to}!=) {
Set(DB(camp/${exten_copy}/call_to)=);
System(call_to ${exten_copy} ${call_to});
}
}
Ah I love to see AEL in a suggestion post :)
--
Sherwood McGowan
VoIP / Telecom Solutions
[EMAIL PROTECTED]
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[asterisk-users] Problem with several includes in ARA
Hi, We are implementing the asterisk realtime architecture using extensions_table, sip_buddies and voicemail_users. We have a problem to make several includes in the ddbb. Only the first include is loaded and the others no. In the following example, only the include lookupdundi is included and not the outbound and applications. +--+--+---+--++-+ | id | context | exten | priority | app| appdata | +--+--+---+--++-+ | 3558 | internal | include |4 | include| lookupdundi | | 3559 | internal | include |5 | include| outbound| | 3560 | internal | include |6 | include| applications| We are working with Asterisk 1.4.18 and MySQL How can it be solved? Cheers, Manuel ___ -- 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 several includes in ARA
On Tue, 2008-06-10 at 18:41 +0200, Samael - wrote: We are implementing the asterisk realtime architecture using extensions_table, sip_buddies and voicemail_users. We have a problem to make several includes in the ddbb. As far as I know, includes are *not* supported at all in the Asterisk Realtime Architecture. (I personally think they should be, but to be honest I'm not a huge fan of ARA anyway, so my opinion probably doesn't count for much.) One workaround is to put your includes in extensions.conf, like this: [internal] include = lookupdundi include = outbound include = applications I know, it's not ideal, but it seems to get the job done. -- Jared Smith Training Manager Digium, Inc. ___ -- 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] SayNumber while reading DTMF?
I'm using the SayNumber() app to read out a users balance for an IVR. Is there a way I can do that while waiting for DTMF input? Obviously, read() and Background() don't correctly say a number in number format. Thanks, Doug. ___ -- 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] SayNumber while reading DTMF?
On Tue, 2008-06-10 at 10:03 -0700, Douglas Garstang wrote: I'm using the SayNumber() app to read out a users balance for an IVR. Is there a way I can do that while waiting for DTMF input? Obviously, read() and Background() don't correctly say a number in number format. I don't know of an easy way of doing this, short of writing a routing in an AGI script to read numbers in the proper format. I'd personally love to see SayDigitsBackground, SayNumberBackground, SayAlphaBackground, etc. or even better, the ability to put numbers in the filename parameters to the Read application, such as: Read(some-variable,your-account-balanceis%123%dollars) (Obviously I just chose the percent sign as an arbitrary delimiter there, but you get the idea.) -- Jared Smith Training Manager Digium, Inc. ___ -- 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] SayNumber while reading DTMF?
Douglas Garstang wrote: I'm using the SayNumber() app to read out a users balance for an IVR. Is there a way I can do that while waiting for DTMF input? Obviously, read() and Background() don't correctly say a number in number format. I do not know of a way to do that. It would be an extremely useful new feature to have, but as fair as I know, is not currently available. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- 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] Weird one way Audio situation
Hi list, I'm having trouble with calls placed to the PSTN (through a TDM card), sometimes (a lot indeed) when I dial a number the callee party can't hear me at all. My setup is: Asterisk 1.4.20.1 Zaptel 1.4.11 libpri 1.4.4 Wanpipe 3.2.4 I have a Sangoma Remora Card A400D (2 FXS / 10 FXO), Grandstream GXP-2000 IP Phones, SuSE Linux Enterprise Server 10 (SP2) x86_64 with Kernel 2.4.16.60-0.23-smp I'm using the ulaw audio codec. There is no NAT between the Asterisk Server and the Phones (the phone and the server are in the same network segment). What can it be??? Thanks in advance for any help/comment... -- Raul Linux Counter #156439 ___ -- 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] Fax on FXS
On Tue, Jun 10, 2008 at 11:15:49AM -0500, Eric ManxPower Wieling wrote: IP can be run over many things. Internet, Local LAN, Corporate WAN, VPN, etc Each of these things have different characteristics and so I add them to the list. FaxOverVoiceOverIPOverLAN is something that has a good chance of working, as a LAN tends to have little latency and little jitter, where FaxOverVoiceOverIPOverInternet has neither low latency nor low jitter. How would suggest I indicate global internet instead of IP on a local lan? Well, I think I'd write it as FAX/VoIP/Internet, but I agree entirely with your semantics. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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] SayNumber while reading DTMF?
Poo. Thanks Jared. - Original Message From: Jared Smith [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 10, 2008 10:24:46 AM Subject: Re: [asterisk-users] SayNumber while reading DTMF? On Tue, 2008-06-10 at 10:03 -0700, Douglas Garstang wrote: I'm using the SayNumber() app to read out a users balance for an IVR. Is there a way I can do that while waiting for DTMF input? Obviously, read() and Background() don't correctly say a number in number format. I don't know of an easy way of doing this, short of writing a routing in an AGI script to read numbers in the proper format. I'd personally love to see SayDigitsBackground, SayNumberBackground, SayAlphaBackground, etc. or even better, the ability to put numbers in the filename parameters to the Read application, such as: Read(some-variable,your-account-balanceis%123%dollars) (Obviously I just chose the percent sign as an arbitrary delimiter there, but you get the idea.) -- Jared Smith Training Manager Digium, Inc. ___ -- 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 ___ -- 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] SayNumber while reading DTMF?
On Tue, Jun 10, 2008 at 01:24:46PM -0400, Jared Smith wrote: (Obviously I just chose the percent sign as an arbitrary delimiter there, but you get the idea.) Oh, ghod; let's not get *that* argument started again... Cheers, -- jr ':-)' a -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- 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 open source codec and sample size
Probably for the same reason that every popular piece of software can be found on torrents with serials and cracks, as well as hundreds if not thousands of sites that just offer serials or cracks to make demo software fully functional. I am not saying I agree with it but it is extremely common. Personally I would love to see Speex as an industry standard. Thanks, Steve Totaro On Tue, Jun 10, 2008 at 12:19 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: The G729 codec is neither open source, nor is it free, and the license/patent does not make an exception for educational use. The Intel LIBRARIES are free for educational/personal use, but the license for that software says that you still need a license from the G729 patent holder before use. I don't understand why people won't pay $10/channel for a fully licensed, legal, and Asterisk supported G729 codec. Manoj_Rajkarnikar wrote: Greetings. I'm new to the asterisk voip world and I'm currently trying out trixbox 2.6.0.7 on a p4 1.8 GHz box. I've downloaded and used the open source g729 codec from site http://asterisk.hosting.lv/ and is working fine. question here is that this codec sends out a packet every 20ms. Though the speech quality is very good, I also like to try out 30ms sampling size to bring down the overhead payload and reduce bandwidth usage. I've searched for it for a couple days with no indication of how to do it. is it possible to change it. do i have to compile my own codec module.. or some patch to asterisk code?? Please suggest. Thanks a lot. Manoj -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 ___ -- 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] SayNumber while reading DTMF?
10 jun 2008 kl. 19.28 skrev Russell Bryant: Douglas Garstang wrote: I'm using the SayNumber() app to read out a users balance for an IVR. Is there a way I can do that while waiting for DTMF input? Obviously, read() and Background() don't correctly say a number in number format. I do not know of a way to do that. It would be an extremely useful new feature to have, but as fair as I know, is not currently available. The ugly way is to use an externail app, like Sox, to concatenate the audio files and then use background(). /O ___ -- 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] Fax on FXS
Find (voice) Replace (Analog) for starters. Bottom line, without writing a book, is that fax and modem (analog) converted to IP (and back) at some point will be altered. There are a multitude of reasons that the audio is altered and to what degree is going to determine your success or failure. FaxOverVoiceOverIPOverInternet has neither low latency nor low jitter is not a provable fact. I have very low latency or jitter to many destinations. It depends how many hops and how much traffic. A routetrace and ping can help you in those regards. Thanks, Steve Totaro On Tue, Jun 10, 2008 at 12:15 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: IP can be run over many things. Internet, Local LAN, Corporate WAN, VPN, etc Each of these things have different characteristics and so I add them to the list. FaxOverVoiceOverIPOverLAN is something that has a good chance of working, as a LAN tends to have little latency and little jitter, where FaxOverVoiceOverIPOverInternet has neither low latency nor low jitter. How would suggest I indicate global internet instead of IP on a local lan? Matt Watson wrote: Ah, you got me there! Could start throwing in a lot of Over's going down that road :) -- Matt http://www.mattgwatson.ca -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Tuesday, June 10, 2008 4:10 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Fax on FXS On Mon, Jun 09, 2008 at 06:53:34PM -0400, Matt Watson wrote: On June 9, 2008 01:34:31 pm Eric ManxPower Wieling wrote: You should not expect FaxOverVoiceOverIPOverInternet to work well. If you stick to ulaw codec for the entire call, it might work well enough for your use, but it might not. Just as an FYI - you have too many Over's in your description FaxOverVoiceOverIP would make sense, but seeing as how IP is short for Internet Protocol, saying Internet Protocol Over Internet doesn;t make much sense... Unless you use an openvpn / ipsec tunnel :-) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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 ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 ___ -- 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] handling jabber status
Thanks for the snippet, I re-wrote it (badly) for regular extensions.conf usage, and verified it's also working here on 1.6, though I do get a warning about JabberStatus being depreciated. Yes, JabberStatus is being moved from an dialplan application to a function (JABBER_STATUS), because it's just retrieving a variable. Philippe ___ -- 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] Polycom SIP and DHCP problem
Lyndon Griffin [EMAIL PROTECTED] writes: I will believe it's a code problem - I see that the phones are picking up *some* of the attributes I pass in DHCP offers, like the domain name. Not only that, but I've sniffed a phone actually trying to ARP the address the server DHCPOFFERED to it, before it decides to use a 192.168.0.x address. This is probably a stupid suggestion, but nevertheless. You have authoritative; in your dhcpd.conf, right? /Benny ___ -- 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 open source codec and sample size
Eric ManxPower Wieling wrote: I don't understand why people won't pay $10/channel for a fully licensed, legal, and Asterisk supported G729 codec. I wish I could use $10/channel G729 codec from Digium, however, I've been trying to get that codec working on Solaris since v32 of that codec. The codec fails to load no matter what I do, and troubleshooting information from Digium (and the lists) is severly lacking. I do understand that it is unsupported, however, I wonder if the people who build the codec have successfully loaded the module within asterisk on Solaris themselves. If I can get this working we would be buying the digium codes without any questions at all. Just my 0.02c ___ -- 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] Interoffice phone setup
Asterisk gets very upset if it can't lookup the host name associated
with every IP on the system, normally it would use DNS to do this, but
since your Internet connection was down it could not do that.
So to clarify, it not only needs to resolve FQDN's, but do reverse lookups
on ip's as well? I am not sure I noticed this, as the external dns provider
it was using would have no reverse lookup zones for the internal clients?
On an additional note, I have not been able to get onsite yet, but the ISP
repaired
the physical link and the system started working but the inbound sip provider
rang
busy until I ssh'ed in and did a reload from the asterisk console? I thought the
system would re register any connections define with a register = every {n}
seconds
on its own? Is there something I can do to force what a reload did
automatically so if the link disappears it repairs itself on its own?
Thanks!
jlc
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Re: [asterisk-users] g729 open source codec and sample size
On Tue, 10 Jun 2008, Bruce McAlister wrote: Eric ManxPower Wieling wrote: I don't understand why people won't pay $10/channel for a fully licensed, legal, and Asterisk supported G729 codec. I wish I could use $10/channel G729 codec from Digium, however, I've been trying to get that codec working on Solaris since v32 of that codec. The codec fails to load no matter what I do, and troubleshooting information from Digium (and the lists) is severly lacking. I do understand that it is unsupported, however, I wonder if the people who build the codec have successfully loaded the module within asterisk on Solaris themselves. If I can get this working we would be buying the digium codes without any questions at all. And of-course some countries don't honour software patents anyway. This may or may not be right in various peoples eyes, but that's the way it is. It's also nice to have a try before you buy too. And there might just be a case where you can't connect an asterisk box to the public Internet to register the licenses (I had that with HPEC some time back) Nothing to stop people wanting to clear their conscious by using the free one and paying for Digium licenses of-course, even if they're not actually used.. Just my 0.02c Euro cents going by the email address :) Gordon ___ -- 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] Interoffice phone setup
The external DNS server would immediately return with a not found
message. Without internet access you'll have to wait for the timeouts, etc.
Joseph L. Casale wrote:
Asterisk gets very upset if it can't lookup the host name associated
with every IP on the system, normally it would use DNS to do this, but
since your Internet connection was down it could not do that.
So to clarify, it not only needs to resolve FQDN's, but do reverse lookups
on ip's as well? I am not sure I noticed this, as the external dns provider
it was using would have no reverse lookup zones for the internal clients?
On an additional note, I have not been able to get onsite yet, but the ISP
repaired
the physical link and the system started working but the inbound sip provider
rang
busy until I ssh'ed in and did a reload from the asterisk console? I thought
the
system would re register any connections define with a register = every
{n} seconds
on its own? Is there something I can do to force what a reload did
automatically so if the link disappears it repairs itself on its own?
Thanks!
jlc
___
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--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network design. Based near
Birmingham, AL. Now accepting clients worldwide.
___
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[asterisk-users] Blind transfers and ringback tone
Hello everyone. I'm having a minor problem with blind transfers and would like to know if it's possible to solve this. Here's the scenario: 1) A receive a call from B 2) A hits flash and gets a dialtone. B is listening music on hold 3) A dials C 4) As soon as C starts ringing, A hangs up. At this point, B listens nothing (mute) 5) When C answers, B and C talk normally The problem is B thinks the call was disconnected because he doesn't listen to the ringback tone when the call is being transferred to C. Here's the CLI output. Considering A = 2001; B = 2000; C = 2002 Connected to Asterisk 1.4.13 currently running on pabx (pid = 4173) Verbosity is at least 3 [Jun 10 16:35:42] -- Executing [EMAIL PROTECTED]:1] Macro(SIP/200-b6f175c0, ramalzap|interna|1|2001) in new stack [Jun 10 16:35:42] -- Executing [EMAIL PROTECTED]:1] GotoIf(SIP/200-b6f175c0, 1?interna) in new stack [Jun 10 16:35:42] -- Goto (macro-ramalzap,s,4) [Jun 10 16:35:42] -- Executing [EMAIL PROTECTED]:4] Ringing(SIP/200-b6f175c0, ) in new stack [Jun 10 16:35:42] -- Executing [EMAIL PROTECTED]:5] Dial(SIP/200-b6f175c0, Zap/1r5|60) in new stack [Jun 10 16:35:42] -- Called 1r5 [Jun 10 16:35:42] -- Zap/1-1 is ringing [Jun 10 16:35:43] -- Zap/1-1 is ringing [Jun 10 16:35:44] -- Zap/1-1 is ringing [Jun 10 16:35:48] -- Zap/1-1 is ringing [Jun 10 16:35:48] -- Starting simple switch on 'Zap/2-1' [Jun 10 16:35:49] -- Zap/1-1 is ringing [Jun 10 16:35:52] -- Hungup 'Zap/2-1' [Jun 10 16:35:52] NOTICE[4998]: cdr.c:434 ast_cdr_free: CDR on channel 'Zap/2-1' not posted [Jun 10 16:35:52] -- Zap/1-1 answered SIP/200-b6f175c0 [Jun 10 16:35:55] -- Started three way call on channel 1 [Jun 10 16:35:55] -- Started music on hold, class 'default', on SIP/200-b6f175c0 [Jun 10 16:35:55] -- Starting simple switch on 'Zap/1-1' [Jun 10 16:35:55] -- Stopped music on hold on SIP/200-b6f175c0 [Jun 10 16:35:55] -- Started music on hold, class 'default', on SIP/200-b6f175c0 [Jun 10 16:35:58] -- Executing [EMAIL PROTECTED]:1] Macro(Zap/1-1, ramalzap|interna|2|2002) in new stack [Jun 10 16:35:58] -- Executing [EMAIL PROTECTED]:1] GotoIf(Zap/1-1, 1?interna) in new stack [Jun 10 16:35:58] -- Goto (macro-ramalzap,s,4) [Jun 10 16:35:58] -- Executing [EMAIL PROTECTED]:4] Ringing(Zap/1-1, ) in new stack [Jun 10 16:35:58] -- Executing [EMAIL PROTECTED]:5] Dial(Zap/1-1, Zap/2r5|60) in new stack [Jun 10 16:35:58] -- Called 2r5 [Jun 10 16:35:58] -- Zap/2-1 is ringing [Jun 10 16:35:58] -- Zap/2-1 is ringing [Jun 10 16:35:59] -- Zap/2-1 is ringing [Jun 10 16:36:00] WARNING[5002]: chan_zap.c:775 zt_get_index: Unable to get index, and nullok is not asserted [Jun 10 16:36:00] -- Hungup 'SIP/200-b6f175c0MASQ' [Jun 10 16:36:00] -- SIP/200-b6f175c0 requested special control 17, passing it to Zap/2-1 [Jun 10 16:36:00] -- Stopped music on hold on Zap/1-1ZOMBIE [Jun 10 16:36:00] -- Hungup 'Zap/1-1' [Jun 10 16:36:00] == Spawn extension (macro-ramalzap, s, 5) exited non-zero on 'Zap/1-1ZOMBIE' in macro 'ramalzap' [Jun 10 16:36:00] == Spawn extension (macro-ramalzap, s, 5) exited non-zero on 'Zap/1-1ZOMBIE' [Jun 10 16:36:03] -- Zap/2-1 is ringing [Jun 10 16:36:04] -- Zap/2-1 is ringing [Jun 10 16:36:05] -- Zap/2-1 answered SIP/200-b6f175c0 [Jun 10 16:36:09] -- Hungup 'Zap/2-1' [Jun 10 16:36:09] == Spawn extension (macro-ramalzap, s, 5) exited non-zero on 'SIP/200-b6f175c0' in macro 'ramalzap' [Jun 10 16:36:09] == Spawn extension (macro-ramalzap, s, 5) exited non-zero on 'SIP/200-b6f175c0' Atenciosamente, Vinícius Fontes Núcleo de Tecnologias Convergentes Canall Tecnologia em Comunicações Passo Fundo - RS - Brasil +55 54 2104-7000 Convergent Technologies Core Canall Tecnologia em Comunicações Passo Fundo - RS - Brazil +55 54 2104-7000 ___ -- 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] Zaptel config
If I am not using any additional hardware and only need ztdummy, would it be sufficient to run make menuconfig and remove all modules except ztdummy or are there additional ones aside from the obvious ones used for hardware I don't have? Given I only have sip voip providers and all my phones are sip based ip phones is there a better way to prevent the unneeded modules from attempting to load at startup? Thanks! jlc ___ -- 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] zaptel issue
Yesterday, freshly out of Asterisk Bootcamp, I attempted to resume an Asterisk installation on a new server. Zaptel 1.4.10.1 had been installed, but I decided to uninstall, and install Zaptel 1.4.11 before I went further ___ -- 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] SayNumber while reading DTMF?
Run a script before the user gets to Background that cat the gsm files
together and then play that file.
IE
#!/bin/bash
BALANCE=$1
ACCOUNT=$2
SOUNDSDIR=/var/lib/asterisk/sounds
ACCOUNTFILE=$SOUNDSDIR/accounts/$ACCOUNT.gsm
#
#Some creative scripting will need to be done to be able to properly say
the #numbers. Ie One-Hundred Eighteen Dollars and Forty-Two Cents. This
script #will play One-One-Eight Dollars and Four-Two cents.
#
# Get the Dollars and Cents...
#
DOLLARS=`echo $BALANCE | cut -f1 -d.`
CENTS=`echo $BALANCE | cut -f2 -d.`
#
#
ELEMENTS=`echo $DOLLARS | wc -m`
for (( i=0;i$ELEMENTS;i++)); do
cat $SOUNDSDIR/digits/${DOLLARS:$i:1}.gsm $ACCOUNTFILE
done
cat $$SOUNDSDIR/dollars.gsm $ACCOUNTFILE
cat $$SOUNDSDIR/and.gsm $ACCOUNTFILE
ELEMENTS=`echo $CENTS | wc -m`
for (( i=0;i$ELEMENTS;i++)); do
cat $SOUNDSDIR/digits/${CENTS:$i:1}.gsm $ACCOUNTFILE
done
cat $$SOUNDSDIR/cents.gsm $ACCOUNTFILE
--
Then just call the script before the Background with the argumants need
and then play back the generated file...
Alex
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Russell Bryant
Sent: Tuesday, June 10, 2008 1:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SayNumber while reading DTMF?
Douglas Garstang wrote:
I'm using the SayNumber() app to read out a users balance for an
IVR.
Is there a way I can do that while waiting for DTMF input?
Obviously, read() and Background() don't correctly say a number in
number format.
I do not know of a way to do that. It would be an extremely useful
new
feature to have, but as fair as I know, is not currently available.
--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.
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Re: [asterisk-users] Polycom SIP and DHCP problem
Occam's Razor wins again! Your assertion that something stupid was to blame was a big help. For posterity, always make sure that some junior admin hasn't used a home router/gateway as an emergency hub stuffed underneath somebody's desk. Those pesky extra DHCP servers don't play nice with others. Thanks Benny Amorsen wrote: Lyndon Griffin [EMAIL PROTECTED] writes: I will believe it's a code problem - I see that the phones are picking up *some* of the attributes I pass in DHCP offers, like the domain name. Not only that, but I've sniffed a phone actually trying to ARP the address the server DHCPOFFERED to it, before it decides to use a 192.168.0.x address. This is probably a stupid suggestion, but nevertheless. You have authoritative; in your dhcpd.conf, right? /Benny ___ -- 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 ___ -- 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] SayNumber while reading DTMF?
Well, how about using an app like cepstral to record it as a wav and using background or waitexten to play the wav -- the time lag should never be noticed. on Tuesday 06/10/2008 Russell Bryant([EMAIL PROTECTED]) wrote Douglas Garstang wrote: I'm using the SayNumber() app to read out a users balance for an IVR. Is there a way I can do that while waiting for DTMF input? Obviously, read() and Background() don't correctly say a number in number format. I do not know of a way to do that. It would be an extremely useful new feature to have, but as fair as I know, is not currently available. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- 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] zaptel issue
Interesting, the bottom of my previous email disappeared ... so here is again. Yesterday, freshly out of Asterisk Bootcamp, I attempted to resume an Asterisk installation on a new server. Zaptel 1.4.10.1 had been installed, but I decided to uninstall, and install Zaptel 1.4.11 before I went further. I get past make clean, ./configure, then when I get to make install I get this error. #$ sudo make install make[1]: Entering directory `/var/ports/zaptel-1.4.11' make -C /lib/modules/2.6.18-53.1.19.el5/build ARCH=x86_64 SUBDIRS=/kernel HOTPLUG_FIRMWARE=yes KBUILD_OBJ_M=pciradio.o tor2.o torisa.o wcfxo.o wct1xxp.o wctdm.o wcte11xp.o wcusb.o zaptel.o ztd-eth.o ztd-loc.o ztdummy.o ztdynamic.o zttranscode.o wct4xxp/ wctc4xxp/ xpp/ wctdm24xxp/ wcte12xp/ modules make[2]: Entering directory `/usr/src/kernels/2.6.18-53.1.19.el5-x86_64' scripts/Makefile.build:17: /kernel/Makefile: No such file or directory make[3]: *** No rule to make target `/kernel/Makefile'. Stop. make[2]: *** [_module_/kernel] Error 2 make[2]: Leaving directory `/usr/src/kernels/2.6.18-53.1.19.el5-x86_64' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/var/ports/zaptel-1.4.11' make: *** [all] Error 2 -Original Message- From: Eve-Ellen Cole [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 10, 2008 4:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: zaptel issue Yesterday, freshly out of Asterisk Bootcamp, I attempted to resume an Asterisk installation on a new server. Zaptel 1.4.10.1 had been installed, but I decided to uninstall, and install Zaptel 1.4.11 before I went further. ___ -- 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 open source codec and sample size
On Tue, Jun 10, 2008 at 08:32:15PM +0100, Gordon Henderson wrote: On Tue, 10 Jun 2008, Bruce McAlister wrote: Eric ManxPower Wieling wrote: I don't understand why people won't pay $10/channel for a fully licensed, legal, and Asterisk supported G729 codec. I wish I could use $10/channel G729 codec from Digium, however, I've been trying to get that codec working on Solaris since v32 of that codec. The codec fails to load no matter what I do, and troubleshooting information from Digium (and the lists) is severly lacking. I do understand that it is unsupported, however, I wonder if the people who build the codec have successfully loaded the module within asterisk on Solaris themselves. If I can get this working we would be buying the digium codes without any questions at all. And of-course some countries don't honour software patents anyway. This may or may not be right in various peoples eyes, but that's the way it is. Most of those countries still honour copyrights. Specifically the copyrights to Intel's IPP code that is used in this codec. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] zaptel issue
On Tue, Jun 10, 2008 at 04:00:44PM -0400, Eve-Ellen Cole wrote: Yesterday, freshly out of Asterisk Bootcamp, I attempted to resume an Asterisk installation on a new server. Zaptel 1.4.10.1 had been installed, but I decided to uninstall, and install Zaptel 1.4.11 before I went further No need to uninstall. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Zaptel config
On Tue, Jun 10, 2008 at 01:48:46PM -0600, Joseph L. Casale wrote: If I am not using any additional hardware and only need ztdummy, would it be sufficient to run make menuconfig and remove all modules except ztdummy or are there additional ones aside from the obvious ones used for hardware I don't have? Given I only have sip voip providers and all my phones are sip based ip phones is there a better way to prevent the unneeded modules from attempting to load at startup? You only need the modules ztdummy and zaptel . Of the utilities you don't even really need ztcfg, but zttest can be handy. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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 : using setvar with IP Realtime and variable inheritance
I did, as a test, and it did work. The problem if that since the SIP phone
(Polycom in my case) is handling the transfer, I have nowhere to put this
line.
What I did, which I thought was the same, as put the underscores in the SIP
registrations table's setvar column. But THAT didn't work.
I`ll take a look at the other solution this evening. Hopefully it's not as
complicated as it looks at first glance.
Regards,
Mick
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Leif Madsen
Sent: Tuesday, June 10, 2008 09:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk : using setvar with IP Realtime and
variable inheritance
Mike wrote:
If I hardcode this value in my dialplan using two underscores before it
(i.e
Setvar(__did=551234) ) this works. But I can't hardcode it, I need
to
fetch it from the table.
Have you tried:
Set(__did=${did})
That might work.
--
Leif Madsen
http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk
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Re: [asterisk-users] g729 open source codec and sample size
On Tue, 2008-06-10 at 20:10 +0100, Bruce McAlister wrote: I wish I could use $10/channel G729 codec from Digium, however, I've been trying to get that codec working on Solaris since v32 of that codec. The codec fails to load no matter what I do, and troubleshooting information from Digium (and the lists) is severly lacking. I see that Jason Parker from Digium answered your question in both July and August of last year. The issue (at least from what I read in the archives) seems to point to math libraries not being found in the proper location. Maybe there are some Solaris folks lurking on the list that can shed some light -- I'm pretty worthless when it comes to Solaris. Are you still trying on OpenSolaris, and is there anything different about the way it handles dynamic linking? I do understand that it is unsupported, however, I wonder if the people who build the codec have successfully loaded the module within asterisk on Solaris themselves. Absolutely! No only have we successfully loaded the module within Asterisk, we've made calls through the system using the g.729 codec to make sure it's actually working. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel issue
On Tue, Jun 10, 2008 at 04:20:32PM -0400, Eve-Ellen Cole wrote: Interesting, the bottom of my previous email disappeared ... so here is again. Yesterday, freshly out of Asterisk Bootcamp, I attempted to resume an Asterisk installation on a new server. Zaptel 1.4.10.1 had been installed, but I decided to uninstall, and install Zaptel 1.4.11 before I went further. I get past make clean, ./configure, then when I get to make install I get this error. #$ sudo make install make[1]: Entering directory `/var/ports/zaptel-1.4.11' make -C /lib/modules/2.6.18-53.1.19.el5/build ARCH=x86_64 SUBDIRS=/kernel HOTPLUG_FIRMWARE=yes KBUILD_OBJ_M=pciradio.o tor2.o torisa.o wcfxo.o wct1xxp.o wctdm.o wcte11xp.o wcusb.o zaptel.o ztd-eth.o ztd-loc.o ztdummy.o ztdynamic.o zttranscode.o wct4xxp/ wctc4xxp/ xpp/ wctdm24xxp/ wcte12xp/ modules make[2]: Entering directory `/usr/src/kernels/2.6.18-53.1.19.el5-x86_64' scripts/Makefile.build:17: /kernel/Makefile: No such file or directory make[3]: *** No rule to make target `/kernel/Makefile'. Stop. Described here, including workaround: http://bugs.digium.com/12750 Any idea when this one last worked? make[2]: *** [_module_/kernel] Error 2 make[2]: Leaving directory `/usr/src/kernels/2.6.18-53.1.19.el5-x86_64' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/var/ports/zaptel-1.4.11' make: *** [all] Error 2 -Original Message- From: Eve-Ellen Cole [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 10, 2008 4:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: zaptel issue Yesterday, freshly out of Asterisk Bootcamp, I attempted to resume an Asterisk installation on a new server. Zaptel 1.4.10.1 had been installed, but I decided to uninstall, and install Zaptel 1.4.11 before I went further. ___ -- 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 -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Seeking Collaboration in Development and Validation of an Anomaly Detection System for Asterisk
We are currently doing research and development on an open-source runtime application monitoring system for Asterisk. This system is aimed at detecting and mitigating problems or vulnerabilities that arise from residual errors--whether unintentional or malicious--either in the application code or in its configuration or usage patterns. It can, for example, be used to detect and prevent various security, performance, and availability problems resulting from latent errors in Asterisk code or, more importantly, in the dialplans it is configured with for handling all calls that go through it. Our approach involves examining events that get generated as a side effect of normal call processing and analyzing them, or some appropriate transformations of those events, against normal, expected application behavior. Certain expected behaviors may be specified explicitly by system experts, while others may be learned implicitly by the monitoring system from training data that represents the target Asterisk PBX's normal, intended usage modes. In many instances, problems detected by the monitoring system may also be addressed automatically if the target system also provides appropriate control interfaces. In the case of Asterisk, for example, the Asterisk Manager Interface (AMI) API may be used for both--obtaining application events as well as performing certain mitigation actions. System logs generated by Asterisk may also act as additional sources of application events. We would like to make the resulting monitoring software available as an open source system for others to use, enhance, and experiment with. To do an effective job, however, we would like to partner with some large, existing Asterisk users, who can help us gather real life examples of Asterisk usage against which we can test and evaluate our techniques. This can, obviously, be done in a manner that addresses the privacy and confidentiality concerns of all parties involved. Any names, phone numbers, and URIs, for example, may be masked appropriately in all data that is shared with others. Please let us know if you would like to participate in this effort or if you have any questions in this regard. Any related help/suggestions/pointers would also be greatly appreciated. Thanks. -- Hira Agrawal Telcordia Technologies [EMAIL PROTECTED] ___ -- 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 configuring a PRI...
I'm trying to get a Qwest PRI configured and working with my lab Asterisk server. They said that the switchtype is 5ess and the signaling is pri_cpe. My entries into zaptel.conf are: span=1,0,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us channels=1-23 And my entries in zapata.conf are: language=en context=telco-incoming switchtype=5ess signalling=pri_cpe rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no group = 1 switchtype = 5ess signalling = pri_cpe group = 1 channel = 1-23 I'm not able to make/receive calls, and the error I'm receiving is: [Jun 10 11:32:37] WARNING[31768]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! == Primary D-Channel on span 1 down Qwest says that the PRI is fine. I have a green light on the PRI card. Help! ___ Chris Hoff Telecommunications Administrator SEI LLC Voice +1 701 298 8865 Ext 2189 Mobile +1 701 361 5976 Fax +1 701 298 8860 Email [EMAIL PROTECTED] ___ -- 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] Delaying SIP disconnect after incoming call hangs up?
I'm looking for a way to delay the disconnection of a call to a SIP extension (or pad it with silence) for a few seconds, after an incoming call to that extension hangs up. Rationale: I have an Asterisk PBX (current 1.4.20 codebase), with a Leadtek BVP8051S ATA hooked to an analog phone which has a built-in answering machine. Incoming SIP connections to the appropriate extension are dialed to this SIP ATA, the phone rings, and the answering machine picks up... all as it should be. However, when the caller hangs up, the ATA immediately starts generating a fast-busy disconnect/congestion beeping. The answering machine doesn't recognize this as a hang-up situation (it expects to hear the line go silent) and it keeps recording beeps until its message-length timer expires and it hangs up the line to the ATA. Unfortunately, I can't change the answering machine's behavior, and I don't think it's possible to change the Leadtek BAP8051S to just go silent. So, what I'm hoping, is that there is some way within Asterisk to change the PBX behavior when the incoming call disconnects, so that it can defer sending the disconnect event to the SIP extension for 10 or 15 seconds... enough quiet time for the answering machine to recognize end-of-call and hang up. I think that either sending nothing (no RTP stream) to the SIP extension, or sending silence or comfort noise frames, would work fine. I've looked through the documentation and through a fair bit of the source code, and haven't found anything which actually works. I tried adding an h hangup rule to the dialplan for this extension, with a Wait(10) action, but this seemed to have no effect. Either the h rule isn't working, or the disconnect frame has already been processed and a SIP BYE has been sent. I've only been able to figure out one approach which *may* work... use an h hangup rule for the extension, which runs a DeadAGI() script, which contacts the SIP ATA via its http administrative interface and reboots the ATA (which immediately drops the line). This may very well work, but is about as elegant as a bag-full of wet tree sloths, and I'd like to do a better job than this. Is there any provision in Asterisk for being able to catch the hangup/disconnect of the far end of a connection, and either wait (with no activity) for a fixed period of time, or do the equivalent of a Play() to send the contents of an audio file to the remaining extension (the target of the Dial() in the extension dialplan)? Currently, the SIP extension in question is behind a NAT, and I've set canreinvite=no, so I believe that all of the SIP and RTP traffic is going through Asterisk. It seems to me that it *ought* to be possible for Asterisk to catch the end-of- connection situation and react in some way other than immediately disconnecting the receiving SIP peer, but I'm not sure that any such capability has been implemented. I realize that the outside-the-box answer to this would be Why use an answering machine? Use the PBX voicemail! but that's not entirely desirable in this situation. Since the phone / answering machine is analog, it has no message waiting light available to let us know that a call has come in, and we'd also lose the ability to jump onto a call which is in the process of being recorded. My wife is comfortable with how the existing answering machine system works, and I'd rather present her with an IP-based solution which doesn't change the behavior she's used to... she's not the most technophilic person around. Thanks in advance for any ideas you can throw my way! ___ -- 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] Delaying SIP disconnect after incoming call hangs up?
On Tue, Jun 10, 2008 at 5:28 PM, Dave Platt [EMAIL PROTECTED] wrote: I'm looking for a way to delay the disconnection of a call to a SIP extension (or pad it with silence) for a few seconds, after an incoming call to that extension hangs up. Rationale: I have an Asterisk PBX (current 1.4.20 codebase), with a Leadtek BVP8051S ATA hooked to an analog phone which has a built-in answering machine. Incoming SIP connections to the appropriate extension are dialed to this SIP ATA, the phone rings, and the answering machine picks up... all as it should be. However, when the caller hangs up, the ATA immediately starts generating a fast-busy disconnect/congestion beeping. The answering machine doesn't recognize this as a hang-up situation (it expects to hear the line go silent) and it keeps recording beeps until its message-length timer expires and it hangs up the line to the ATA. Unfortunately, I can't change the answering machine's behavior, and I don't think it's possible to change the Leadtek BAP8051S to just go silent. So, what I'm hoping, is that there is some way within Asterisk to change the PBX behavior when the incoming call disconnects, so that it can defer sending the disconnect event to the SIP extension for 10 or 15 seconds... enough quiet time for the answering machine to recognize end-of-call and hang up. I think that either sending nothing (no RTP stream) to the SIP extension, or sending silence or comfort noise frames, would work fine. I've looked through the documentation and through a fair bit of the source code, and haven't found anything which actually works. I tried adding an h hangup rule to the dialplan for this extension, with a Wait(10) action, but this seemed to have no effect. Either the h rule isn't working, or the disconnect frame has already been processed and a SIP BYE has been sent. I've only been able to figure out one approach which *may* work... use an h hangup rule for the extension, which runs a DeadAGI() script, which contacts the SIP ATA via its http administrative interface and reboots the ATA (which immediately drops the line). This may very well work, but is about as elegant as a bag-full of wet tree sloths, and I'd like to do a better job than this. Is there any provision in Asterisk for being able to catch the hangup/disconnect of the far end of a connection, and either wait (with no activity) for a fixed period of time, or do the equivalent of a Play() to send the contents of an audio file to the remaining extension (the target of the Dial() in the extension dialplan)? Currently, the SIP extension in question is behind a NAT, and I've set canreinvite=no, so I believe that all of the SIP and RTP traffic is going through Asterisk. It seems to me that it *ought* to be possible for Asterisk to catch the end-of- connection situation and react in some way other than immediately disconnecting the receiving SIP peer, but I'm not sure that any such capability has been implemented. I realize that the outside-the-box answer to this would be Why use an answering machine? Use the PBX voicemail! but that's not entirely desirable in this situation. Since the phone / answering machine is analog, it has no message waiting light available to let us know that a call has come in, and we'd also lose the ability to jump onto a call which is in the process of being recorded. My wife is comfortable with how the existing answering machine system works, and I'd rather present her with an IP-based solution which doesn't change the behavior she's used to... she's not the most technophilic person around. Thanks in advance for any ideas you can throw my way! Your ATA is to blame. It is generating the noise you describe. Maybe there is a setting on the ATA to address this. Idea, try an Grandstream 286 ATA. I know people bash Granstream and I have in the past as to their phones, but their ATAs are pretty good. Thanks, Steve T ___ -- 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 configuring a PRI...
On Tue, 2008-06-10 at 16:22 -0500, Christopher Hoff wrote: I'm trying to get a Qwest PRI configured and working with my lab Asterisk server. They said that the switchtype is 5ess and the signaling is pri_cpe. Your configuration looks correct to me at first glance. I'm not able to make/receive calls, and the error I'm receiving is: [Jun 10 11:32:37] WARNING[31768]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! == Primary D-Channel on span 1 down Qwest says that the PRI is fine. I have a green light on the PRI card. If you have a green light on the back of the T1 card, then you're at least seeing framing and line-coding from Qwest, which is the first step. The second (and completely different) step is getting the D-channel to come up. What did Qwest say when you told them that you're not seeing the D-channel come up on the circuit? It's been my experience with Qwest PRIs that you have to call them and have them reset a few things on their switch to get the D-channel to come up. -- Jared Smith Training Manager Digium, Inc. ___ -- 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 open source codec and sample size
Jared Smith wrote: I see that Jason Parker from Digium answered your question in both July and August of last year. The issue (at least from what I read in the archives) seems to point to math libraries not being found in the proper location. Maybe there are some Solaris folks lurking on the list that can shed some light -- I'm pretty worthless when it comes to Solaris. Are you still trying on OpenSolaris, and is there anything different about the way it handles dynamic linking? Yes, Jason answered the question saying that the codec was unsupported and the other suggestion that was given was that it could possibly be that the license was in the wrong directory. This is the first time that I've heard of the math library not being in the correct location? Do you have a reference as to what Jason mentioned about the math library? When I first posed the question on the lists and a question via the digium channels I mentioned that I was using Solaris 10 Update 3. Which is what I was told the codec was built on. I've not tried it on OpenSolaris at all. The company I work for will only use the standard Solaris distribution, and not OpenSolaris in production. -- +---+ | Bruce McAlister Blueface Ltd | | [EMAIL PROTECTED] http://www.blueface.ie | +---+ ___ -- 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 configuring a PRI...
1. Ditch the channels= in zaptel.conf that doesnt belong there (you've done the channel config with the bchan= and dchan= 2. your span= should *probably* be 1,1 instead of 1,0 in zaptel.conf the 2nd 1 indicates to use that span as a primary timing source 3. not that it should matter, but you don;t need the duplicate group=, signalling=, switchtype= in zapata.conf 4. you can ditch rxwink= that setting is for non-PRI T1s try that and see if that helps... I suspect the span not being used as primary timing source is whats causing your greif. good luck! -- Matt Watson http://www.mattgwatson.ca On June 10, 2008 05:22:40 pm Christopher Hoff wrote: I'm trying to get a Qwest PRI configured and working with my lab Asterisk server. They said that the switchtype is 5ess and the signaling is pri_cpe. My entries into zaptel.conf are: span=1,0,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us channels=1-23 And my entries in zapata.conf are: language=en context=telco-incoming switchtype=5ess signalling=pri_cpe rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no group = 1 switchtype = 5ess signalling = pri_cpe group = 1 channel = 1-23 I'm not able to make/receive calls, and the error I'm receiving is: [Jun 10 11:32:37] WARNING[31768]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! == Primary D-Channel on span 1 down Qwest says that the PRI is fine. I have a green light on the PRI card. Help! ___ Chris Hoff Telecommunications Administrator SEI LLC Voice +1 701 298 8865 Ext 2189 Mobile +1 701 361 5976 Fax +1 701 298 8860 Email [EMAIL PROTECTED] ___ -- 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 open source codec and sample size
Jared Smith wrote: The issue (at least from what I read in the archives) seems to point to math libraries not being found in the proper location. Maybe there are some Solaris folks lurking on the list that can shed some light -- I'm pretty worthless when it comes to Solaris. Are you still trying on OpenSolaris, and is there anything different about the way it handles dynamic linking? I forgot to mention, in my previous email, that the math libraries on our boxes reside in the /lib directory, which is where the Solaris installer installs them by default. Looking at my last attempt to try and get this going (which, co-incidently, is the same system that Jason helped me with) I checked to see if the codec has any unresolved libraries: ldd ./codec_g729a.so libgcc_s.so.1 = /usr/sfw/lib/libgcc_s.so.1 libc.so.1 = /lib/libc.so.1 libm.so.2 = /lib/libm.so.2 The math libraries appear to be found OK on the box. The license is located in : /var/lib/asterisk/licenses The license file is in the directory: -rw-r--r-- 1 root root 308 Aug 27 2007 G729-39F0ABB3.lic However, every time I try to load the codec, I get the following in the asterisk console: codec_g726.so = (ITU G.726-32kbps G726 Transcoder) [Jun 10 23:16:40] NOTICE[2673]: codec_g729.c:403 load_module: G.729 transcoding module version 32, Copyright (C) 1999-2007 Digium, Inc. [Jun 10 23:16:40] NOTICE[2673]: codec_g729.c:407 load_module: This module is supplied under a commercial license granted by Digium, Inc. [Jun 10 23:16:40] NOTICE[2673]: codec_g729.c:408 load_module: Please see the full license text supplied by the accompanying [Jun 10 23:16:40] NOTICE[2673]: codec_g729.c:409 load_module: register utility, or ask for a copy from Digium. [Jun 10 23:16:40] NOTICE[2673]: codec_g729.c:410 load_module: This product includes software developed by the OpenSSL Project [Jun 10 23:16:40] NOTICE[2673]: codec_g729.c:411 load_module: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [Jun 10 23:16:40] NOTICE[2673]: codec_g729.c:412 load_module: Copyright (C) 1998-2006 The OpenSSL Project [Jun 10 23:16:40] WARNING[2673]: codec_g729.c:420 load_module: Failed to initialize G.729 copy protection! codec_g729a.so = (Annex A/B (floating point) G.729 Codec (optimized for i386)) In this case I am using asterisk v1.4.13, however, I have tried this with asterisk versions: 1.2.17 - 29 1.4.13 - 18 The codec versions I have tried are the i386 32-bit below: unsupported v32 unsupported v33 unsupported trunk v33 I cannot seem to locate version 34 for Solaris on the download site which is apparently the latest version which I have not tried as of yet. When I built asterisk I changed the directory locations to install everything in /opt/asterisk as apposed to spread over multiple directories. This would be the ideal case for us. However, when trying to get it to work as expected, I built asterisk using the default install directories to rule out any weirdness I may have caused by modifying the make file to install to a single top level directory. I've also asked the guys at SolarisVoIP some time ago to see if they had got G729 going, and as far as I am aware, they have not been able to get the codec working either on their Solaris systems. There are multiple posts on that mailing list where people mention large scale rollouts on Solaris being held back because they are unable to get the G729 codec operational under Solaris. I am not alone :) Any suggestions tips/tricks that you may be able to shed on this issue would be *greatly* appreciated. Thanks Bruce ___ -- 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 open source codec and sample size
[Jun 10 23:16:40] WARNING[2673]: codec_g729.c:420 load_module: Failed to initialize G.729 copy protection! Even though unrelated to Solaris, I have seen this exact same error on Linux when the License is invalid/outdated. In our specific case we had a very old box with a 2004 license. We upgraded everything to 1.4.20 including the G729 binary. When starting asterisk with the old license it spit out the above error. Deleting the old license and running the latest register utility fixed the issue. codec_g729a.so = (Annex A/B (floating point) G.729 Codec (optimized for i386)) This line would seem to indicate the binary loads fine. I would concentrate on the License aspect. Delete the license from the directory and see if you get the same 'copy protection error'. If not it means the License location was correct but the file has a problem. Andres http://www.neuroredes.com ___ -- 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] Polycom SIP and DHCP problem
For posterity, always make sure that some junior admin hasn't used a home router/gateway as an emergency hub stuffed underneath somebody's desk. Those pesky extra DHCP servers don't play nice with others. Just a suggestion, will defining a special VLAN just for the Polycom phone be able to get rid of such potential problems? ___ -- 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 configuring a PRI...
Here is my configuration with Global Crossing. Hope this helps. Zaptel.co # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 B8ZS/ESF ClockSource span=1,1,0,esf,b8zs # termtype: te bchan=1-23 dchan=24 Zapata.conf mode=mixed signalling=pri_cpe context=incoming-att group=1 channel = 1-23 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Hoff Sent: Tuesday, June 10, 2008 5:23 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Problems configuring a PRI... I'm trying to get a Qwest PRI configured and working with my lab Asterisk server. They said that the switchtype is 5ess and the signaling is pri_cpe. My entries into zaptel.conf are: span=1,0,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us channels=1-23 And my entries in zapata.conf are: language=en context=telco-incoming switchtype=5ess signalling=pri_cpe rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no group = 1 switchtype = 5ess signalling = pri_cpe group = 1 channel = 1-23 I'm not able to make/receive calls, and the error I'm receiving is: [Jun 10 11:32:37] WARNING[31768]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! == Primary D-Channel on span 1 down Qwest says that the PRI is fine. I have a green light on the PRI card. Help! ___ Chris Hoff Telecommunications Administrator SEI LLC Voice +1 701 298 8865 Ext 2189 Mobile +1 701 361 5976 Fax +1 701 298 8860 Email [EMAIL PROTECTED] ___ -- 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] Sound files custom path
I need to play sound files located outside default asterisk directory. Is there a way to specify full path? Regards MR ___ -- 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 Installation with Radius Support
Isn't this exactly why lists such as asterisk-users evolve? I don't see anything wrong with his question, and I see full connection to Asterisk to boot! Nothing OT about it either. So, why, if I may, does he have to ask a smart question, and/or advertise on asterisk-biz? Did the 'step by step' lead you to such a conclusion? Don't even know where you get the notion that he wants a consultant? Let's start using forums, lists, and IRC to help please. Your answer sounds just like one of those pricks on IRC saying Go read the book. for any question they get. Thanks, Mark. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Griffin Sent: June 9, 2008 2:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Installation with Radius Support Abid Saleem wrote: Hi All, Can someone provide me a step by step guide to install and configure Asterisk 1.2 with Radius using agi scripts. I have currently installed andconfigured it but it is not disconnecting the call after the credit_time returned by radius. So I am guessing I may have missed some configuraton. Please help. You may find it useful to read http://catb.org/~esr/faqs/smart-questions.html Alternatively, you may want to advertise on asterisk-biz, to look for a consultant. -- Give me fruitful error any time, full of seeds, bursting with its own corrections. You can keep your sterile truth for yourself. - Vifredo Pareto ___ -- 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 ___ -- 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] Sound files custom path
rossi.tek wrote: I need to play sound files located outside default asterisk directory. Is there a way to specify full path? Sure... just specify the full path.. exten = s,1,Playback(/path/to/my/file) -- Leif Madsen http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- 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] Sound files custom path
On Tuesday 10 June 2008 21:41:29 Leif Madsen wrote: rossi.tek wrote: I need to play sound files located outside default asterisk directory. Is there a way to specify full path? Sure... just specify the full path.. exten = s,1,Playback(/path/to/my/file) Just remember to leave off the suffix (.gsm, .ul, .wav, etc.). -- Tilghman ___ -- 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 Installation with Radius Support
Here is info on Asterisk and Radius http://www.voip-info.org/wiki/view/CW+Radius++for+Asterisk -E ___ -- 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 configuring a PRI...
On Tue, Jun 10, 2008 at 04:22:40PM -0500, Christopher Hoff wrote: I'm trying to get a Qwest PRI configured and working with my lab Asterisk server. They said that the switchtype is 5ess and the signaling is pri_cpe. My entries into zaptel.conf are: span=1,0,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us channels=1-23 Remove this line, please . channels is a valid zaptel.conf line, but will configure those channels for use with pciradio. IIRC 'channels=1-23' is an invalid line. And my entries in zapata.conf are: language=en context=telco-incoming switchtype=5ess signalling=pri_cpe rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no group = 1 switchtype = 5ess signalling = pri_cpe group = 1 channel = 1-23 I'm not able to make/receive calls, and the error I'm receiving is: [Jun 10 11:32:37] WARNING[31768]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! == Primary D-Channel on span 1 down Qwest says that the PRI is fine. I have a green light on the PRI card. What do you see in 'cat /proc/zaptel/*' ? Have you run ztcfg after editing zaptel.conf ? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] TE110P with 40,000 IRQ missess
I have an Asterisk server that was running fine until Sunday. Monday there was a power outage and the server was off most of the day. This server has a TE110P, two TDM04B and an Astribank 32. Today I noticed that the TE110P started having IRQ missess. Before today it only had about two or three per month. Today, this is the miss count after ten minutes on: IRQ Misses: 1443 I can see the number climbing when using zttool. Users are reporting noise and echo on the analog channels. Obviously faxes are not going through anymore which has the client up in arms. I think thet maybe the card got damaged? I can send and receive calls on the E1 link and users with SIP phones do not seem affected (at least they are not the ones complaining). The TE110P is not sharing and IRQ with anything else. Any ideas on where to look for the problem? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ -- 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] TE110P with 40,000 IRQ missess
If you don't have a spare card, try resetting the PCI bus in the Bios, it may have become corrupt with the power failure. At least try a different slot. You can also try flashing the BIOS. That is the only thing that comes to mind at this time and not knowing if you have a spare card. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Carlos Chavez Sent: Wednesday, June 11, 2008 1:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] TE110P with 40,000 IRQ missess I have an Asterisk server that was running fine until Sunday. Monday there was a power outage and the server was off most of the day. This server has a TE110P, two TDM04B and an Astribank 32. Today I noticed that the TE110P started having IRQ missess. Before today it only had about two or three per month. Today, this is the miss count after ten minutes on: IRQ Misses: 1443 I can see the number climbing when using zttool. Users are reporting noise and echo on the analog channels. Obviously faxes are not going through anymore which has the client up in arms. I think thet maybe the card got damaged? I can send and receive calls on the E1 link and users with SIP phones do not seem affected (at least they are not the ones complaining). The TE110P is not sharing and IRQ with anything else. Any ideas on where to look for the problem? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ -- 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 ___ -- 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
