RE: [Asterisk-Users] SNMP Monitoring
Hi, -Original Message- I use MRTG to graph Active/Configured SIP channels and Active/Total PRI/ZAP channels, but I don't monitor the up/down status. You could probably Any chance you will share the mrtg setup you used for that ? How did you read out asterisk (via manager interface, tailing logfiles, or ... ?) How busy is your setup ? Thanks, Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voice Quality
I would use g.729, and if this is an issue, GSM. Setup trunking between both IAX peers so that you can save a lot of bandwidth. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, 4 May 2005 00:52 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voice Quality Hello, I have setup two * servers and they are communicating using IAX. I'm passing calls from SRV A (internet connection T1) to SRV B (internet connection: 512). For some reasons I have an issue with the quality. The voice is a bit scratchy. I have tried iLBC and SPEEX, but it didn't make any difference. Now, assuming that I have an issue with Bandwidth, what would be the best way to configure my iax.conf. (A bit confused about jitterbuffer and tos) Here is my iax.conf @ location A: [general] port=4569 bandwidth=low disallow=all allow=ilbc ;allow=ulaw ;allow=speex jitterbuffer=200 jitterbuffer=yes tos=lowdelay and iax.conf @ location B: [general] port=4569 bandwidth=low disallow=all allow=ilbc ;allow=ulaw ;allow=speex jitterbuffer=200 jitterbuffer=yes tos=lowdelay [guest] type=user context=default callerid=Guest IAX User disallow=all allow=ilbc Thanks guys ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored... Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Therefore, we do not accept responsibility for any errors or omissions that are present in this message, or any attachment, that have arisen as a result of e-mail transmission. If verification is required, please request a hard-copy version. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CODEC Allow statement help
Hello, I have 6 Asterisk switches all running together nicely with DUNDi and have one minor problem with inter switch CODEC negotiation. I use G729 (licensed from Digium) on several of the switches. Inbetween the G729 switches we can make calls no problem. From a switch that only does ULAW they cannot make a call into my G729 switch. (the call fails with an RTP translation error) The G729 switch can build a call toward the ULAW switch, and the call is processed as ULAW. On my G729 switches: both sip/iax.conf show disallow=all allow=g729 allow=ulaw On the ULAW switches: disallow=all allow=ulaw I have tried every odd combination of allow statements for the ULAW switches to build a call while still allowing the G729 switches to build calls. The phones are a mixture of Budgettones and Polycom IP600's. The phones all have thier first codec set to g729. the second to ULAW. My question is this, in a multi CODEC environment where some phones are ULAW, some are G729, and some switches are ULAW, some are G729 licensed, what is the best set of statements to get them all to play together? thanks! Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Freak incidents, who's to blame?
On Tue, 3 May 2005, Andrew Kohlsmith wrote: On May 3, 2005 02:22 pm, Ryan Courtnage wrote: From what I've read, glare is common in 2-way loopstart (kewlstart) circuits, and is impossible(?) to eliminate completely. But now I'm wondering what Nortel would tell a customer who experiences glare on their new Meridian system... they must do something to prevent glare from happening. Any ideas? Nope. Technically it shouldn't be possible with PRI but it is and does happen. Typically you hunt up starting at the highest available channel, and the telco hunts down which tends to keep it at bay until things get busy. Glare is when both the net and the cpe end attempt to seize a line simultaneously and both believe they succeeded. Glare really is impossible on a pri as a B channel can not be requested and allocated to both parties by mistake. The handshaking performed leaves no ambiguity as to which call a line is allocated to. However, a similar situation can occur when the cpe end requests a specific B channel in a SETUP message instead of leaving the channel selection to the net end. Unlike the glare condition this situation is detected and the net end prevails. The cpe end should then try to allocate another B channel with a new SETUP message. Unfortunatly Asterisk as a cpe device neither lets the net end allocate the B channel, nor does it retry using a different B channel. The problem is that Asterisk does not see the whole PRI as a single link with several channels, it sees the inidvidual channels with a common signalling path. A specific B channel is allocated before the signalling starts. This is a deficiency in Asterisk, not in isdn in general. The solution for Asterisk is the same as for glare-prone links - hunt for channels in the opposite direction. Note that on isdn links quite a few operators will by default _not_ hunt from one end or another, this has to be requested. The convention then is for the net end to hunt low-to-high and the cpe end to hunt high-to-low. Finally, even on isdn you have end devices (phones) which may themselves be prone to the human equivalent of glare - picking up the handset before the ring is heared. Some phones allow the user to request an outside line by pressing a button to prevent this. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 compile error.
Hi. I downloaded pwlib_1.18.1 and openh323_1.15.1 to install Asterisk CVS HEAD version. I tried to install asterisk-oh323-0.7.1. I patched openh323 as typing 'patch -p1 /usr/src/asterisk-oh323-0.7.1/openh323_1.13.5-make.patch' in openh323 directory. Then I compiled pwlib, openh323 and installed Asterisk. After that, I edited 'Makefile' in asterisk-oh323-0.7.1 directory. I typed 'make', and there is an error. [EMAIL PROTECTED] make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk-oh323-0.7.1/wrapper' ./check_ver /root/work/pwlib pwlib ./check_ver /root/work/openh323 openh323 g++ -DP_USE_PRAGMA -D_REENTRANT -Wall -fPIC -I/root/work/pwlib/include -DPTRACING -I/usr/local//include/openh323 -DHAS_OSS -DHAS_VPB -Wall -x c ++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.8.1\ -DOPENH323VERSION=\1.15.1\ -I/root/work/pwlib/include/ptlib/unix -I/root/work/pwlib/include -I/root/work/openh323/include -I/root/work/openh323/include/openh323 -I../asterisk-driver -c wrapper_misc.cxx -o wrapper_misc.o touch ../asterisk-driver/chan_oh323.c g++ -DP_USE_PRAGMA -D_REENTRANT -Wall -fPIC -I/root/work/pwlib/include -DPTRACING -I/usr/local//include/openh323 -DHAS_OSS -DHAS_VPB -Wall -x c ++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.8.1\ -DOPENH323VERSION=\1.15.1\ -I/root/work/pwlib/include/ptlib/unix -I/root/work/pwlib/include -I/root/work/openh323/include -I/root/work/openh323/include/openh323 -I../asterisk-driver -c asteriskaudio.cxx -o asteriskaudio.o touch ../asterisk-driver/chan_oh323.c g++ -DP_USE_PRAGMA -D_REENTRANT -Wall -fPIC -I/root/work/pwlib/include -DPTRACING -I/usr/local//include/openh323 -DHAS_OSS -DHAS_VPB -Wall -x c ++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.8.1\ -DOPENH323VERSION=\1.15.1\ -I/root/work/pwlib/include/ptlib/unix -I/root/work/pwlib/include -I/root/work/openh323/include -I/root/work/openh323/include/openh323 -I../asterisk-driver -c wrapconnection.cxx -o wrapconnection.o touch ../asterisk-driver/chan_oh323.c g++ -DP_USE_PRAGMA -D_REENTRANT -Wall -fPIC -I/root/work/pwlib/include -DPTRACING -I/usr/local//include/openh323 -DHAS_OSS -DHAS_VPB -Wall -x c ++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.8.1\ -DOPENH323VERSION=\1.15.1\ -I/root/work/pwlib/include/ptlib/unix -I/root/work/pwlib/include -I/root/work/openh323/include -I/root/work/openh323/include/openh323 -I../asterisk-driver -c wrapendpoint.cxx -o wrapendpoint.o wrapendpoint.cxx: In member function `virtual BOOL WrapH323EndPoint::OpenAudioChannel(H323Connection, int, unsigned int, H323AudioCodec)': wrapendpoint.cxx:915: error: `IsDescendant' undeclared (first use this function) wrapendpoint.cxx:915: error: (Each undeclared identifier is reported only once for each function it appears in.) make[1]: *** [wrapendpoint.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.1/wrapper' make: *** [subdirs_build] Error 1 [EMAIL PROTECTED] Please let me know to solve that problem. Thanks for reading. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk CDR - Mysql
We seem to be having the same problem. The cdr command is not found, so we tried to do a make and install on the add-ons but it can't see to find the files when we run 'make clean make make install'. We have downloaded from CVS and the files look to be there but it still can't find the files. Could someone help? Thanks Rick [EMAIL PROTECTED] asterisk-addons]# make clean make make install rm -f *.so *.o .depend make -C format_mp3 clean make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' rm -f *.o *.so *~ make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` app_addon_sql_mysql.c:31:19: mysql.h: No such file or directory cdr_addon_mysql.c:33:19: mysql.h: No such file or directory cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o common.o common.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o dct64_i386.o dct64_i386.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o decode_ntom.o decode_ntom.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o layer3.o layer3.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o tabinit.o tabinit.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o interface.o interface.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o format_mp3.o format_mp3.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -shared -Xlinker -x -o format_mp3.so common.o dct64_i386.o decode_ntom.o layer3.o tabinit.o interface.o format_mp3.o make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' for x in format_mp3/format_mp3.so ; do install -m 755 $x /usr/lib/asterisk/modules ; done [EMAIL PROTECTED] asterisk-addons]# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Monday, May 02, 2005 7:03 PM To: Asterisk Users Subject: Re: [Asterisk-Users] Asterisk CDR - Mysql If you enabled it in logger.conf, it should be at /var/log/asterisk/debug What does cdr mysql status do? If it says no such command then you haven't loaded the cdr module. Did you do make install inside the asterisk-addons dir? Do you have autoload = yes in your modules.conf? -Matthew From: Callum McGillivray [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tue, 03 May 2005 12:03:17 +1000 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk CDR - Mysql Which Debug log ? Where is it located ? I can't see anything obvious that shows this info. Cheers, Callum (P.S. I'm not seeing a connection on the mySQL DB from the asterisk machine, and I assumed that there should be one... what am I missing here ? ) Matthew Boehm wrote: What is in your debug log? It will show the exact SQL that is being executed. -Matthew From: Callum McGillivray [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tue, 03 May 2005 11:35:52 +1000 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk CDR - Mysql Hi All, We have configured our Asterisk Server (CVS Head) to use mysql for CDR's, following the guidelines located at http://www.voip-info.org/wiki-Asterisk+cdr+mysql . When Asterisk starts up there are no errors, when we make a call there are no errors, however I am not seeing records in the database. Any idea how what I should be looking for here? I'm a bit lost. Cheers, Callum ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com
[Asterisk-Users] Asterisk and Post Paid Billing
Could somebody recommend a good software utility, preferably with a web front, end for post paid billing in Asterisk? I've seen a lot of discussion on the various pre-paid and calling card based solutions, but nothing that would allow me to configure different regex-based locations/costs and generate a bill for a given user (users sorted out by SetAccount app and resulting application codes appended to CDR) at the END of the month. Thank you, EZ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNMP Monitoring
Hi, We use Cacti (an MRTG based monitoring tool), and I would also like to see how you set that up. Any chance you are willing to share ? Cheers, Callum Florian Overkamp wrote: Hi, -Original Message- I use MRTG to graph Active/Configured SIP channels and Active/Total PRI/ZAP channels, but I don't monitor the up/down status. You could probably Any chance you will share the mrtg setup you used for that ? How did you read out asterisk (via manager interface, tailing logfiles, or ... ?) How busy is your setup ? Thanks, Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with TDM400P card
On Tue, 2005-05-03 at 07:39 -0600, Rich Adamson wrote: To help identify the source of the delays, I built a new system this weekend from scratch. When that is complete, I'll use it to compare the differences in motherboards, OS distro's, and maybe kernel versions. Very good Rich, the results of that work will be very interesting. And now for the results (thus far) Built a new system from scratch using ECS PM800-M2 Mobo, ide 7200 rpm drive, 2.7ghz celery, 512 meg, fedora 3 (v2.6.9-1.667, no updates). With TDM04b installed only (new system): - 'vmstat 1' shows 100% cpu every 8 seconds with no significant changes while processing a single pstn or iax call. - zttest shows 99.987793% consistently with no significant variation - wctdm using Int #11 (no sharing) Well, if you would like another data point, my current system has: 1 x X100p 1 x TDM40b (I think, quad FXS) 1 x TE410p (Quad E1 card) it never has 0% under the idle column, though it does occassionally approach 50% (eg, 52% etc) and is a dual AMD Athlon MP CPU. Results from zttest (while asterisk is running): --- Results after 63 passes --- Best: 100.00 -- Worst: 99.987793 All samples are 100% except 13 which are 99.987793 All cards are on their own IRQ (16/17 and 18) Kernel 2.6.11 (plain linux kernel, custom compiled) More details available on request, just let me know what you want... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dial analog phone with sip
i ,sorry but i've tried to made my extensions like you but nothing .now i've fxs card and i can recieve calls in my analog card from sip but i can not dial out anaother analog phone . please help me Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails !Créez votre Yahoo! Mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk to analog pbx
Hi
this is the macro used for that purpose ..
[macro-dialout-trunk]
exten = s,1,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4);check
for CID override for exten
exten = s,2,SetCallerID(${ECID${CALLERIDNUM}})
exten = s,3,Goto(6)
exten = s,4,GotoIf($[foo${OUTCID_${ARG1}} = foo]?6);check
for CID override for trunk
exten = s,5,SetCallerID(${OUTCID_${ARG1}})
exten = s,6,SetGroup(OUT_${ARG1})
exten = s,7,CheckGroup(${OUTMAXCHANS_${ARG1}})
; if we've used up the max channels, continue at 108 (n+101)
exten = s,8,SetVar(DIAL_NUMBER=${ARG2})
exten = s,9,SetVar(DIAL_TRUNK=${ARG1})
exten = s,10,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the proper
dial string for this trunk
exten = s,11,Dial(${OUT_${ARG1}}/${OUTPREFIX_${ARG1}}${DIAL_NUMBER})
; if dial fails (ie, all channels are busy), continue at 112 (n+101)
;exten = s,11,Dial(Zap/0/${DIAL_NUMBER})
; we should only get here if the call was successful (?)
exten = s,9,Congestion
; exit points for macro
exten = s,108,NoOp(max channels used up)
exten = s,112,NoOp(dial failed)
as u can see is also a dial instruction
the call seems to be done but in fact my analog extension does not
ring :/
any clue?
Thanks again
El mar, 03-05-2005 a las 10:17 -0500, Moises Silva escribió:
Hi Julio. It would be nice if you show the extensions.conf that
handles that kind of calls. You can do something like this:
[macro-analogpbx]
exten = s,1,Cut(ChannelType=CHANNEL,/,1) //check if the call comes
from other Zap ch
exten = s,2,GotoIf($[${ChannelType} = Zap] ? 3 : 6) //If does, go 3,
othewise 6
exten = s,3,Flash()
exten = s,4,SendDTMF(${analogprefix}${num}) //send the DTMF for the
extension dialed
exten = s,5,Hangup()
exten = s,6,Dial(Zap/g${analoggroup}/${analogprefix}${num}) //if the
call comes from SIP or IAX then execute Dial trough some group in
zapata
exten = s,7,Hangup()
You can see some variables i just use for administration of my PBX,
but i hope you understand the concept.
Good Look
- moy
On 5/3/05, Julio Saura [EMAIL PROTECTED] wrote:
Hi there
i have an asterisk box running ok, and now i am trying to integrate it
with my local analog pbx
So far, i have connected the fxo port of my * to an analog extension
port of my analog pbx.
As far as i know, if a call an extension of my analog pbx on a sip phone
( i have done the right dial plan for routing these calls to de zap
channel ) the analog pbx extension should ring ...
am i right?
asterisk says the call is done, but the analog extension keeps in
silence .. :?
any clue, am i doing something wrong?
Best regards.
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Re: [Asterisk-Users] TDM users: modified zttest.c for testing
I also had problem faxing with spandsp with my old server (Athlon 700 on a VIA chipset). Now I've instaled asterisk on a P4 2.8Ghz (Asus P5P800, btw great board, let's you assign the preferred interrupt for each PCI slot), with 256Mb, and here's what I get (unpatched zttest): (before I never got to 100%) [EMAIL PROTECTED] zaptel]# ./zttest -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% --- Results after 19 passes --- Best: 100.00 -- Worst: 99.987793 I have yet to try spandsp, but I think i'll work without problems. Julian J. M. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Quality
What's your end device? if it's a voip device (eg SIP phone or a soft phone) then you shouldn't need a jitter buffer. Also, you don't need bandwidth=low if you specify the codecs (the disallow=all will override the bandwidth=low) and maxjitterbuffer is the param you're after with this line jitterbuffer=200 I'm guessing -Adam [EMAIL PROTECTED] wrote: Hello, I have setup two * servers and they are communicating using IAX. I'm passing calls from SRV A (internet connection T1) to SRV B (internet connection: 512). For some reasons I have an issue with the quality. The voice is a bit scratchy. I have tried iLBC and SPEEX, but it didn't make any difference. Now, assuming that I have an issue with Bandwidth, what would be the best way to configure my iax.conf. (A bit confused about jitterbuffer and tos) Here is my iax.conf @ location A: [general] port=4569 bandwidth=low disallow=all allow=ilbc ;allow=ulaw ;allow=speex jitterbuffer=200 jitterbuffer=yes tos=lowdelay and iax.conf @ location B: [general] port=4569 bandwidth=low disallow=all allow=ilbc ;allow=ulaw ;allow=speex jitterbuffer=200 jitterbuffer=yes tos=lowdelay [guest] type=user context=default callerid=Guest IAX User disallow=all allow=ilbc Thanks guys ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk to analog pbx
Hello all is right, the analog extension should ring, but maybe your dialplan is not correct or you call a bad extension in you PBX. can you post your dialplan?, to see it. regards - Original Message - From: Julio Saura [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, May 03, 2005 2:37 PM Subject: [Asterisk-Users] asterisk to analog pbx Hi there i have an asterisk box running ok, and now i am trying to integrate it with my local analog pbx So far, i have connected the fxo port of my * to an analog extension port of my analog pbx. As far as i know, if a call an extension of my analog pbx on a sip phone ( i have done the right dial plan for routing these calls to de zap channel ) the analog pbx extension should ring ... am i right? asterisk says the call is done, but the analog extension keeps in silence .. :? any clue, am i doing something wrong? Best regards. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mysql/Radius Authentication
Hi all, I'm using asterisk-1.0.7. I need to configure asterisk in such way that it authenticates users from mysql DB. Is it possible to authenticate SIP users from mysql database? It seems to me that chan_sip2 code from Olle E. Johansson, Edvina.net, [EMAIL PROTECTED] can authenticate users from mysql. However I looked for it everywhere and didn't find. Where can I download chan_sip2 code? Is there any other way I can authenticate SIP users from mysql in asterisk? Is it possible to make asterisk work with radius? I appreciate if somebody can give me some hints and advices in this regard. thanks in advance, Ganbold ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phones for home use?
On 03/05/05, Justin B Newman [EMAIL PROTECTED] wrote: Neil Cherry wrote: What are your recommendations for a slightly fancy home phone? The Sipura SPA-841 is a nice compromise between the Ciscos and the Grandstreams. Check out the new Grandstream GXP-2000. I've been testing these, they're much better than the BT-100s. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk to analog pbx
Hi i posted it this morning i guess is a [EMAIL PROTECTED] problem... installing a new OS with * from scratch it does not even call outside connecting fxo to pots :? El mié, 04-05-2005 a las 09:55 +0200, Mehdi Chouikh escribió: Hello all is right, the analog extension should ring, but maybe your dialplan is not correct or you call a bad extension in you PBX. can you post your dialplan?, to see it. regards - Original Message - From: Julio Saura [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, May 03, 2005 2:37 PM Subject: [Asterisk-Users] asterisk to analog pbx Hi there i have an asterisk box running ok, and now i am trying to integrate it with my local analog pbx So far, i have connected the fxo port of my * to an analog extension port of my analog pbx. As far as i know, if a call an extension of my analog pbx on a sip phone ( i have done the right dial plan for routing these calls to de zap channel ) the analog pbx extension should ring ... am i right? asterisk says the call is done, but the analog extension keeps in silence .. :? any clue, am i doing something wrong? Best regards. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detecting Fax and bad CDRs
On Tue, 2005-05-03 at 13:41 -0500, Matthew Boehm wrote: Personally, I presume you would need to bill your user for that 15 seconds, or else you will end up losing money. You're exactly right. Not only that, but if the call is Answer'd() by asterisk, the disposition becomes 'ANSWERED' in the cdr AND I now have billable seconds. That's right, so you will pay for the call... This is bad 'cause if someone calls a customer and hangs up before VM picks up, the call was Answered, the CDR shows billable time and therefore CDR will tell my customer the call was answered, instead of missed and I bill them for that. Well, the customer's phone never answered the call, so it should still show as a missed call. Unless you get the missed calls list from the CDR... Oh, and wouldn't you HAVE to bill them, or else you are losing money, possibly a lot? Is there any way to change the CDR info after the callee has hung up? The only way to accomplish what you would (AFAICT) would be to make some new app_unanswer or something, which would mark somewhere within asterisk, that the call has not been answered, even though physically/protocol on the card, the line has been answered. Overall, I'd say that even this wouldn't work easily, since it would probably confuse chan_sip or other channels, when the other end is sending audio, and we don't expect that on a un-answered call... Just my thoughts... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bad CLI colors? bad terminal?
On Tue, 2005-05-03 at 20:05 -0500, Matthew Boehm wrote: Holy crap! You mean someone actually read my email? Thanks Andrew. Wish more people would read emails. Just read it :) I run from safe_asterisk and have the line ASTARGS=-n in it. Because I too hate the changing background. WFM(tm) -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Difference between Asterisk and Asterisk@home?
Hi, can one summarize the main differences between Asterisk and [EMAIL PROTECTED] or point me to a location where i can find such a list? Much thanks, Kib ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Audio quality problem recording calls using gsm codec
In article [EMAIL PROTECTED], xlab [EMAIL PROTECTED] wrote: When using phones that are using G.711 codec and the calls are recorded with Monitor, when played back the files sound great. When we use gsm codec at one or both ends of the call, the recorded files sound very bad. Much worse than the audio sounds during the call. With the Monitor command we have tried WAV, wav, and gsm and this does not make any noticable difference, the sound quality is still poor (actually about the same each way). This is probably because Asterisk calls sox to mix the separate incoming and outgoing files into a single file. In order to mix two gsm files, sox will need internally to convert them both to linear, do the mixing, and then convert back to gsm. Since gsm is not a lossless compression, the sound gets worse with each conversion round-trip. I'm not sure what you can do about it. Try wav again, which is supposed to be linear. WAV and gsm are both GSM compressed. Or possibly you could try signed linear explicity as a format (can't remember whether it is sln or slin). Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE:oh323 compile error
Hi Try the step descibed at this link: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg86875.html and make attention to edit correctly Makefile. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: TDM users: modified zttest.c for testing
On Tue, 2005-05-03 at 21:28 +, Tony Mountifield wrote: I've done a few more tests and think I may have uncovered a problem in the pseudo-driver. Whether it's relevant to Rich's problem I don't know, but it might have something to do with MeetMe drift on SIP channels. I modified Rich's program as follows: a) Changed char buf[8192]; to char buf[SIZE]; (SIZE is 8000). b) Changed the objective line to: printf(Objective: to read %d bytes from TDM card in 1.00 seconds.\n, sizeof(buf)); On running it again I was surprised to find it STILL showing times for 8192 bytes instead of 8000. I added the following line just after the read() in the main loop: printf(\nread(fd, buf, %d) returns %d, sizeof(buf), res); That showed me that read(fd,buf,8000) was returning 1024 bytes. Aha, so by the time count = SIZE, it had read 8 blocks totalling 8192. I changed the read to res=read(fd,buf,sizeof(buf)-count) so it would stop at 8000 bytes, and got the following results: Funny I tried much the same thing this morning before reading your results, and was amazed to find the same. Whether you used 8192 or 8000 the time was the same. But as I've posted earlier faxes do work perfectly for me, though the traffic is negligible. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Collect calls
Since you are referring to R2 signaling, it works like this: The E1 R2 Call Blocking feature provides two ways to block incoming collect calls-category-based and double answer. With category-based call blocking, collect calls will be blocked based on a specific category. For example, in Brazil, collect calls arrive with a category II-8, for which the gateway should send B-7 as a response instead of an answer signal. This approach is only applicable when switches in the central office support category-based blocking. For legacy switches that do not support category-based blocking, the double answer method is implemented to support the collect-call blocking. For an incoming collect call, the gateway will answer the call with a clearback after one second and re-answer the call after two seconds, causing the collect call to be dropped and normal calls to stay connected. This is what the referenced patches are attempting to do. This does not work in the U.S. or if you have SS7, you don't need it. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael D Schelin Sent: Tuesday, May 03, 2005 6:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Collect calls You Bring up a great point. I understand these codes and my system brings them in via ss7 but as youself I don't know how to protect my network from these charges. I will follow this post to see if anybody has a fix. Rodrigo P. Telles wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Folks, Does someone knows how to identify and block collect calls on Asterisk using PRI channels? I googled it and found this: http://lists.digium.com/pipermail/asterisk-dev/2004-November/007500.html I don't know what does it mean!!! Can someone help me to understand this? I tried to apply that way too, using Flash() but Flash() complains and looks like just work with FXO channels: http://lists.digium.com/pipermail/asterisk-users/2004-October/066360.html Thanks in advance. - -- Rodrigo P. Telles [EMAIL PROTECTED] IVOZ # 1009 TI Manager Devel-IT - http://www.devel.it Bestcom Group -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFCd+zUiLK8unYgEMQRAkChAJ4xDYOvl8yZY+Uqn6v5VFZ4tMzicQCfT8+T 5foewh0m/o3ABMqcNHhtQs4= =rsu2 -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Data calls trough IAX?
Hello, I have 2 *, one is between a Siemens HiPath and the PSTN, having two PRIs connected to each side. When I call the Hipath to administer it (with Siemens HiPath Manager), I usually call through the PSTN and all wents well. However, I have a second Asterisk and when I call the first Asterisk trough the second to connect to the HiPath, the call comes not through. To show you what I mean: This works: HiPath Manager -- ISDN PBX -- PSTN -- Asterisk1 -- HiPath This doesn't work: HiPath Manager -- ISDN PBX -- Asterisk2 -- Asterisk1 -- HiPath Note: Voice calls are working perfectly, it's only the data calls that doesn't work. The debug output shows the following: -- Accepting unauthenticated call from XXX.XXX.XX.XX, requested format = 8, actual format = 8 -- Executing Dial(IAX2/[EMAIL PROTECTED]/1, Zap/g1/12345678) in new stack -- Called g1/12345678 -- Executing Dial(Zap/5-1, Zap/g2/12345678) in new stack -- Making new call for cr 32776 Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 2 (reference 8/0x8) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [...] -- Channel 0/1, span 2 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request I think the problem is the transfer capability: Speech line. It must be transfer capability: Unrestricted digital information. Is there a way to set the transfer capability? I noticed there is a file app_settransfercapability.c in CVS (but not in 1.0.7). Is this possible with IAX at all? Regards, Henry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Difference between Asterisk and Asterisk@home?
Kib Eki wrote: Hi, can one summarize the main differences between Asterisk and [EMAIL PROTECTED] or point me to a location where i can find such a list? See [EMAIL PROTECTED] site - http://asteriskathome.sf.net - and asterisk site - www.asterisk.org. Basically, asterisk is a program, and [EMAIL PROTECTED] is a distribution with running (and partially configured) asterisk, AMP, etc. and other additional stuff. Of course you have to configure your asterisk hardware yourself. It's like a question: what's the difference between KDE and Debian.. :) Tomek -- Znajdz swoja milosc na wiosne... http://link.interia.pl/f187a ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bristuff-RC8b-CVS
for anyone is using RC8b-CVS: there are some major bugs in asterisk chan_sip and utils. It's convenient to download new asterisk/utils.c and asterisk/channels/chan_sip.c and reapply the kapejod patches to chan_sip.c ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap (or carrier) issue ?
Hi ! I'm a very happy user of Asterisk for my work since a few weeks now, and I have almost everything working perfectly. I can get calls from our 3 T0 France Telecom lines, dial all SIP phones and queues internally or externally, and also dial all national numbers. My main problem is that I can't seem to be able to dial out anything except the national numbers. This works : Executing Dial(SIP/200-0b73, ZAP/g1/155171587) in new stack This doesn't : Executing Dial(SIP/200-b132, ZAP/g1/0033155171587) in new stack Trace : VERBOSE[17251]: -- Called g1/0033155171587 VERBOSE[17211]: -- Channel 0/1, span 1 got hangup I have tried with any number of prefixed zeros (from 0 to 4), with the same result. I also have the same problem with the special french numbers like 1016 for the pro support line of France Telecom, or anything that's not a national number (prefix, then 8 numbers). I'm using an OctoBRI card with bristuff 0.20-RC7k and asterisk 1.0.6. Since everything else is working correctly I guess there is something I do not know about these numbers using a France Telecom T0 line, so I'm asking you guys with experience if there is a simple (and stupid I guess) solution to this. Thanks in advance for your help, and big thanks to the dev team for this fine piece of software. J-F Mammet jfm(at)telechargement.fr ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: NVBackgroundDetect
Date: Tue, 03 May 2005 23:14:18 -0600 From: Joseph [EMAIL PROTECTED] Subject: [Asterisk-Users] NVBackgroundDetect Is anybody using NVBackgroundDetect to detect fax signal on SIP protocol from ATA? -- #Joseph We're using NVBackgroundDetect with SIP and IAX. Several of our products/customers use it successfully -- and in volume. In fact, I think we've only experienced one or two missed detections, which was most likely configuration. For SIP, we've tested all the Sipura (SPA-1000, SPA-1001, SPA-2000, SPA-2100, and SPA-3000), Mediatrix, D-Link, and Linksys adapters. We've tested several providers and * to * configs. No problems. Make sure you follow the wiki instructions for setup and that you understand usage. Link quality, protocol, provider, and mo/do determine that actual fax send/receive results (beyond detection). Great results here as well on send/receive, except on the software side with PDF and TIFF GPL libraries. T.38 and others attempt to solve FOIP, but G711 is working well for most of our clients. The modules have been tested over LAN, DSL, cable, T1, and OC3. DSL and cable worked well, except in some areas cable had problems. In some situations, DSL performance was worse, but generally it was better. BTW- On another note, Cisco recently purchased Sipura. Hopefully they won't kill the company like they have others. Justin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE:How to use ser with asterisk server for load sharing
Hi friends ! Can anybody help me that how to use ser with asterisk server so that ser can work like the front end of the asterisk and all other features of the asterisk can be used. I have tried the configuration given in asterisk-wiki/at+large but could not succeed, still my asterisk in not listening to ser or ser is not forwarding to asterisk. Thanks Deepak Dhiman -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Thursday, April 28, 2005 12:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RJ45 to RJ11? The RJ11 plug fits perfectly into an RJ45 socket and only cares about the center-most conductors, which are the ones with the connection to the PSTN. Mojo Paul Shiflet wrote: I just received my TDM400 card from digium with 2 fxo and 2 fxs interfaces. They are all RJ45 ports as opposed to RJ11 like my POTS phones. How do i interface my POTS phones with this; can i just crimp an RJ45 connection on the end of the phone cord? Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Freak incidents, who's to blame?
On May 4, 2005 02:54 am, Peter Svensson wrote: Unfortunatly Asterisk as a cpe device neither lets the net end allocate the B channel, nor does it retry using a different B channel. The problem is that Asterisk does not see the whole PRI as a single link with several channels, it sees the inidvidual channels with a common signalling path. A specific B channel is allocated before the signalling starts. This is a deficiency in Asterisk, not in isdn in general. Wow, thank you for this very insightful response... It's concise and describes exactly what the problem is, and why. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] atxfer features in stable release.
Hi all. At the end, i get atxfer with sip dowloading head cvs version of asterisk and this is ok, but now i have errors with h323. following the instructions i could compile h323 channel and load it, but when i call from sip to h323 or viceversa, i obtain this. debug - May 4 12:12:07 WARNING[14186]: chan_h323.c:836 oh323_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4) May 4 12:12:07 WARNING[14186]: chan_h323.c:836 oh323_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4) May 4 12:12:07 WARNING[14186]: chan_h323.c:836 oh323_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4) May 4 12:12:07 WARNING[14186]: chan_h323.c:836 oh323_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4) -- H323/as5300-1.lpa.idec.net answered SIP/u0001-fbca May 4 12:12:07 WARNING[14186]: channel.c:2261 ast_channel_make_compatible: No path to translate from SIP/u0001-fbca(4) to H323/212.xxx.xxx.xxx(256) May 4 12:12:07 WARNING[14186]: app_dial.c:1315 dial_exec_full: Had to drop call because I couldn't make SIP/u0001-fbca compatible with H323/212.xxx.xxx.xxx == Spawn extension (default, 828111044, 1) exited non-zero on 'SIP/u0001-fbca' - end debug in the stable version, all its ok WHEN ATXFER AND THE REST OF FEATURESMAP FEATURES IN THE STABLE RELEASE? Best Regards¡¡¡ César García. Director de Sistemas, IdecNet S.A. Centro de Gestión de Red. Edificio IdecNet. C/Juan XXIII 44. E-35004, Las Palmas de Gran Canaria, Islas Canarias - España. Tfn: +34 828 111 000 Ext: 340 Henry Jensen escribió: Hello, I have 2 *, one is between a Siemens HiPath and the PSTN, having two PRIs connected to each side. When I call the Hipath to administer it (with Siemens HiPath Manager), I usually call through the PSTN and all wents well. However, I have a second Asterisk and when I call the first Asterisk trough the second to connect to the HiPath, the call comes not through. To show you what I mean: This works: HiPath Manager -- ISDN PBX -- PSTN -- Asterisk1 -- HiPath This doesn't work: HiPath Manager -- ISDN PBX -- Asterisk2 -- Asterisk1 -- HiPath Note: Voice calls are working perfectly, it's only the data calls that doesn't work. The debug output shows the following: -- Accepting unauthenticated call from XXX.XXX.XX.XX, requested format = 8, actual format = 8 -- Executing Dial(IAX2/[EMAIL PROTECTED]/1, Zap/g1/12345678) in new stack -- Called g1/12345678 -- Executing Dial(Zap/5-1, Zap/g2/12345678) in new stack -- Making new call for cr 32776 Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 2 (reference 8/0x8) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [...] -- Channel 0/1, span 2 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request I think the problem is the transfer capability: Speech line. It must be transfer capability: Unrestricted digital information. Is there a way to set the transfer capability? I noticed there is a file app_settransfercapability.c in CVS (but not in 1.0.7). Is this possible with IAX at all? Regards, Henry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Freak incidents, who's to blame?
Everyone has probably experienced this at some point in the past: You pick up your analog phone. Rather than hearing dialtone, you are connected with someone who has just called you. Neither you nor them heard a ring. Maybe it's just me, but it seems these freak incidents would occur more frequently years ago, than now. I've now experienced this a couple of times with an * system (TDM400p - quad FXO): A SIP exten dials digits which are answered by a Zap trunk. As soon as Zap answers, the SIP extension is connected with an inbound (PSTN) caller (who was expecting to hear an IVR). My questions are: Who's to blame (telco, tdm card, * config, gremlins)? Is this avoidable? I dont know who to blame, but we've had the same problem here with our small sales team. The sales team (about once a week) will dial a call on their analog phones (analog cordless phones plugged into a few SPA-2001s) - they press 'talk', dial the #, then immediatly are connected to an incomming call... (I use two TDM quad FXO cards to service 8 incomming lines from Sprint). I havnt been able to track it down, and its not reproducable manually... From the description, it almost sounds like glare. With analog fxo lines, that essentially means that both asterisk and the telco central office attempted to use the same pstn line for outgoing and incoming lines at the same time. Statistically, glare will occur more frequently with _small_ numbers of pstn lines and _greater_ amounts of traffic. I'd also guess that part of the problem might relate to how asterisk handles call setup. In other words, when an incoming call arrives at asterisk, asterisk probably doesn't mark the line as busy until after the callerid arrives (and the first internal ring occurs). If an out- going call is initiated at that time, asterisk may not know an incoming call is just arriving. But, that's a guess for sure. Might try using immediate=yes and usecallerid=no to see if that has any impact. If it does, then suspect the above timing issue. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM users: modified zttest.c for testing
On Tue, 2005-05-03 at 09:48 -0600, Rich Adamson wrote: TDM X100P card users: Attached is a modified zaptel/zttest.c app called attest-mod.c. It has been modified to report the delay in receiving 8,192 bytes from the TDM card (instead of reporting a percentage). It works with the digium x100p cards as well. Drop the attachment in your zaptel directory and compile it with: gcc zttest-mod.c -o zttest-mod.o Then run the executable like this: ./zttest-mod.o -v and report the results. The output should look like: 8192 bytes in 1.023843 seconds 8192 bytes in 1.023866 seconds 8192 bytes in 1.023853 seconds 8192 bytes in 1.023876 seconds 8192 bytes in 1.023841 secondsr --- Results after 5 passes --- Best: 1.023876 -- Worst: 1.023841 -- Average: 1.023856 The design objective of the TDM (and x100p) cards was to transfer 8,192 bytes of data from the card in exactly 1.0 seconds. The above sample indicates my system required 1.023856 seconds to accomplish this, or 23856 microseconds too late. Isn't the design objective to read 8000 bytes in one second? The reported (roughly) 1.024 second time frame is correct for 8192 bytes. I get average numbers very close to 1.024 (especially if I take some rounding error into account). That's a very good point. Now I'm not sure since the only thing I've got to go by is existing code in zttest.c which implies 8192, and data arrives in 1024 byte frames. I'll dig a little deeper to see if I can figure out which one _is_ correct. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 Help
I just got Mediatrix 1204 from ebay, but it is missing CD that conmtain the software and drivers, I am wondering if anybody knows where I could downloaded from. The firmware is not openly available. Mediatrix approach is to charge customers for every release they generate, and they only do that through approved resellers. If you know a company that resells their products, you might be able to twist their arm, but I'd guess they aren't going to give it away. (That's probably why it was being sold on eBay in the first place.) You will need the firmware that runs on the box (be sure to get the sip version), and you'll need the Windows-only snmp management software to configure the thing. Each firmware version has a specific snmp management package intended to be used with the firmware. You'll need both (matching) to accomplish anything as there is no telnet or web interface. No no no. Screw windows. All you need is the mib files and mbrowse. SNMP makes remote admin of these boxes a piece of cake. Much faster then a web browser. Once you figure out what you are doing, then you can just config and admin it with simple shell scrips, or if your a hack like me, c code. You can even use SNMP to monitor the PSTN line status. Way cool stuff and these boxes just run forever. For those of us that are somewhat heavy into snmp, I'd agree. But a large percentage of asterisk users don't ever deal with it or even know what it is. I'd agree on the stability of the box. Very nice, good echo cancellation, etc. Less then satisfactory in how they deal with sip (eg, registration), security, etc. For internal use, no problem; for external, I'd never expose it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 'multi-line' configuration
Yes. Quoting Henry Devito [EMAIL PROTECTED]: Are you using asterisk @ home? - Original Message - From: Patrick M. Gray, Jr. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 03, 2005 9:22 PM Subject: Re: [Asterisk-Users] 7960 'multi-line' configuration I still can't get the multi-line magic to happen. When I get the second call, this is what appears on the CLI. Any ideas? Thanks! Pat dialparties.agi: Caller ID is not set -- dialparties.agi: Added extension 200 to extension map -- dialparties.agi: Extension 200 cf is disabled -- dialparties.agi: Extension 200 do not disturb is disabled == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 dialparties.agi: Extension 200 has call waiting disabled dialparties.agi: Max calls of 1 exceeded - deleting from dial dialparties.agi: Dial string is empty - nothing to do dialparties.agi: Was direct call, jumping to priority 23 -- AGI Script Executing Application: (NoOp) Options: () -- AGI Script dialparties.agi completed, returning 0 -- Executing Wait(IAX2/[EMAIL PROTECTED]/23, 1) in new stack -- Executing VoiceMail(IAX2/[EMAIL PROTECTED]/23, [EMAIL PROTECTED]) in new stack -- Playing 'voicemail/default/200/unavail' (language 'en') == Spawn extension (macro-exten-vm, s, 6) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' in macro 'exten-vm' == Spawn extension (ext-local, 200, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Freak incidents, who's to blame?
Everyone has probably experienced this at some point in the past: You pick up your analog phone. Rather than hearing dialtone, you are connected with someone who has just called you. Neither you nor them heard a ring. I don't think this is a freak incident at all. It still happens to me with people I call frequently and is easily explainable. you make a call, the telco connects it, and before the ring generator comes into a phase of putting voltage on the line, they pick up the phone. The circuit was connected, it just never got a chance to ring, there is nothing freak about it, just a matter of timing. Might also add that most central office switches do not sync the ringback audio with the actual ringing of the pstn line. So, ringback in many cases may be several seconds before/after the actual pstn line is ringing. Listening for ringback will not be a valid indicator of anything just in case someone suggests doing that. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: TDM users: modified zttest.c for testing
It would be very interesting to see everyone's results in running this, and even more interesting to report the results with the OS distro in use, mobo in use (if known), etc. If anyone actually get's a result that is very close to 1.000 seconds, I'd really like to know more about those systems. (email off list is fine if you want.) Details below; Fedora Core 1. I also tried using nice to raise the process priority, but it made no difference. [EMAIL PROTECTED] zaptel]# ./zttest-mod -v Objective: to read 8192 bytes from TDM card in 1.00 seconds. Opened pseudo zap interface, measuring accuracy... 8192 bytes in 1.023981 seconds 8192 bytes in 1.023995 seconds 8192 bytes in 1.023992 seconds 8192 bytes in 1.023996 seconds 8192 bytes in 1.023991 seconds 8192 bytes in 1.023994 seconds 8192 bytes in 1.023992 seconds 8192 bytes in 1.023994 seconds 8192 bytes in 1.024003 seconds 8192 bytes in 1.023986 seconds 8192 bytes in 1.023992 seconds 8192 bytes in 1.023993 seconds 8192 bytes in 1.023994 seconds 8192 bytes in 1.023993 seconds 8192 bytes in 1.023993 seconds 8192 bytes in 1.023995 seconds 8192 bytes in 1.023992 seconds 8192 bytes in 1.023995 seconds 8192 bytes in 1.023992 seconds 8192 bytes in 1.023993 seconds --- Results after 20 passes --- Best: 1.024003 -- Worst: 1.023981 -- Average: 1.023993 This looks very close to 1024ms instead of 1000ms. That got me thinking: I believe your premise is wrong. The sample rate of telephony audio is 8kHz. With 8-bit samples (uLaw or aLaw), that means 8000 bytes should be supplied in 1 second, not 8192. At a rate of 8000 bytes/sec, 8192 bytes will arrive in 1.024 seconds. That makes a lot of sense and also supports the reported numbers that folks are posting. Can we actually _assume_ 8000 bytes/sec though? In other words, is there something more in the inbound frame (besides pcm audio) to indicate which of 4 ports the data belongs to, etc? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing calls, X100P
Yes I tried the rx and tx values, but no luck there. Then I removed
everything from this line, adsl, fax, etc. and left only asterisk and
still not working. Then I tried the following to get the dialtone and
dial digits myself
exten = _9,1,Dial(${TRUNK}/)
And that didn't work either. I also added ww in front of the dial
string to have it wait a bit more.
Well at least I know that the system is working on the other line
without any problem. As for this line, although every other regular
phone is able to make calls, asterisk can't and that is due to either
the ADSL service or other wiring problem on the line itself. Next I am
going to check resistance/etc on the line.
Thanks for the help
--
Mehmet
Iain Young wrote:
Hi Mehmet,
On Tue, May 03, 2005 at 11:20:44AM -0400, You wrote:
I tried that and it didn't work. Then I decided to use a different phone
line. I had not thought about this before, it just didn't occur to me.
And everything worked fine. The phone line that doesn't work is my ADSL
line.
Hmm. interesting that my line was also an ADSL line. Did you just try
the values for rx and txgain that I gave you, or did you go higher ? I
started at 10db gain (Horrible echo, but dialed), then I worked down,
until I found the lowest gain that gave me minimal echo, but still worked.
Wall to splitter, one side going to ADSL router the other going
into a fax machine and than from fax machine going into X100P. I
remember seeing a post about this before. I'll have to check into that.
I'd suggest trying it Wall - Splitter - X100P, and see what that does.
Asterisk can act as a fax machine anyway.
All the Best
Iain
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[Asterisk-Users] IPSwitchBoard version 0.113 released
Version 0.113 - 4. may 2005 * Can now transfer recorded conversations to your PC automatically * You can configure a folder to hold recordings * You can now specify that all conversations on an extension should be recorded * It's possible to attach a customised string to the recording file name Download: http://ipswitchboard.thorben.dk ___ IPSwitchBoard is a FREE Windows.Net application that will allow you to: Unattended/attended transfers. Park calls and retrieve/forward them again. Organize all your Zap, SIP and IAX extensions (automatically retrieved from Asterisk). Hotel/Call Shop Billing module Monitor all extensions. Monitor all queues. Monitor Agents. Monitor Parked Calls. Dynamically log extensions in and out of queues. Integration with CRM software on the web. Record conversations. Browse Call Records Drop any active call. Set Do Not Disturb on Extensions and give a reason. Speed Dialling. User selectable ring tones for IPSwitchBoard. User selectable button colors. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: TDM users: modified zttest.c for testing
It would be very interesting to see everyone's results in running this, and even more interesting to report the results with the OS distro in use, mobo in use (if known), etc. If anyone actually get's a result that is very close to 1.000 seconds, I'd really like to know more about those systems. (email off list is fine if you want.) --- Results after 20 passes --- Best: 1.024003 -- Worst: 1.023981 -- Average: 1.023993 This looks very close to 1024ms instead of 1000ms. That got me thinking: I believe your premise is wrong. The sample rate of telephony audio is 8kHz. With 8-bit samples (uLaw or aLaw), that means 8000 bytes should be supplied in 1 second, not 8192. At a rate of 8000 bytes/sec, 8192 bytes will arrive in 1.024 seconds. I've done a few more tests and think I may have uncovered a problem in the pseudo-driver. Whether it's relevant to Rich's problem I don't know, but it might have something to do with MeetMe drift on SIP channels. I modified Rich's program as follows: a) Changed char buf[8192]; to char buf[SIZE]; (SIZE is 8000). b) Changed the objective line to: printf(Objective: to read %d bytes from TDM card in 1.00 seconds.\n, sizeof(buf)); On running it again I was surprised to find it STILL showing times for 8192 bytes instead of 8000. I added the following line just after the read() in the main loop: printf(\nread(fd, buf, %d) returns %d, sizeof(buf), res); That showed me that read(fd,buf,8000) was returning 1024 bytes. Aha, so by the time count = SIZE, it had read 8 blocks totalling 8192. I changed the read to res=read(fd,buf,sizeof(buf)-count) so it would stop at 8000 bytes, and got the following results: [EMAIL PROTECTED] zaptel]# ./zttest-mod -v Objective: to read 8000 bytes from TDM card in 1.00 seconds. Opened pseudo zap interface, measuring accuracy... read(fd, buf, 8000) returns 1024 read(fd, buf, 6976) returns 1024 read(fd, buf, 5952) returns 1024 read(fd, buf, 4928) returns 1024 read(fd, buf, 3904) returns 1024 read(fd, buf, 2880) returns 1024 read(fd, buf, 1856) returns 1024 read(fd, buf, 832) returns 832 8000 bytes in 1.023988 seconds read(fd, buf, 8000) returns 1024 read(fd, buf, 6976) returns 1024 read(fd, buf, 5952) returns 1024 read(fd, buf, 4928) returns 1024 read(fd, buf, 3904) returns 1024 read(fd, buf, 2880) returns 1024 read(fd, buf, 1856) returns 1024 read(fd, buf, 832) returns 832 8000 bytes in 1.023998 seconds --- Results after 2 passes --- Best: 1.023998 -- Worst: 1.023988 -- Average: 1.023993 [EMAIL PROTECTED] zaptel]# So it looks like the pseudo driver is always handling 1024 byte chunks, and even if you ask it for fewer bytes, it takes 1024 bytes' worth of time. I think it should really be handling 1000-byte chunks in 125ms rather than 1024-byte chunks in 128ms, if it is supposed to be emulating telephony channels. But zaptel.c is Deep Magic, and I'd be interested in comments from those who are famliar with it in detail. Tony, that is exactly the same path I was looking at when I modified the zttest.c code. However, it appears I got caught making assumptions relative to 1024 * 8 = 8192 bytes in 1.000 sec when it now appears the correct number really is 1.024 sec. I have to be out of the office today, but will dig into the above tonight to see what can be discovered in zaptel. (I'm not a very proficient coder though.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN transfer, handoff to masterswitch
Hi! I've setup * with a Junghanns.net QuadBRI card agaisnt an Ericsson MD-110 PBX. All four ISDN channels are setup to simulate EuroISDN Point-to-Point (Anlagenanschluss in Germany) from the Ericsson's side. Works well and I have had little problems at all. Now what's happening during a call transfer of an ISDN call to another ISDN destination? Does such a transfer block 2 b-channels (one in and one out)? Or is it automatically handed off to the Ericsson so * is out of the loop? Is EuroISDN capable of such a handoff? Or do I have to use another protocol like Q.SIG for that feature? I don't have experience with Q.SIG at all. I was happy * worked with the EuroISDN trunk so well. But there's the issue with congestioning the link to the PBX with too many (in my case 4) forwarded calls. Does anyone have experiences with this? Are there references for Q-SIG out there? Thanks! Alex Mack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN transfer, handoff to masterswitch
On Wed, 4 May 2005, Alex Mack wrote: I've setup * with a Junghanns.net QuadBRI card agaisnt an Ericsson MD-110 PBX. All four ISDN channels are setup to simulate EuroISDN Point-to-Point (Anlagenanschluss in Germany) from the Ericsson's side. Works well and I have had little problems at all. Now what's happening during a call transfer of an ISDN call to another ISDN destination? Does such a transfer block 2 b-channels (one in and one out)? Or is it automatically handed off to the Ericsson so * is out of the loop? Is EuroISDN capable of such a handoff? Or do I have to use another protocol like Q.SIG for that feature? In EuroISDN this functionallity is provided by ECT (Explicit Call Transfer) for established calls and CD (Call Deflection) for calls in the setup phase. Bristuff supports these according to the documentation. Q.SIG has one (or possibly several) ways of doing this. At least one of these have been implemented in libpri. I don't have experience with Q.SIG at all. I was happy * worked with the EuroISDN trunk so well. But there's the issue with congestioning the link to the PBX with too many (in my case 4) forwarded calls. Does anyone have experiences with this? Are there references for Q-SIG out there? I think at least some of the standards are published at http://www.ecma-international.org/activities/Communications/QSIG_page.htm Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Put a wait in a .call file.
Does anyone know of a way to put a wait or a pause in a .call file? When my * tries to make an outgoing call on a Zap channel, it does not wait for a dialtone. It just starts dialing. Thanks, -Ronan signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GXP-2000 review..
As no-one had actually put any technical details about how things work, I wrote up a review of the GXP-2000 today. http://www.gladstonewireless.net/tiki-index.php?page=GXP-2000 --Rob -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Peter Bowyer Sent: Wednesday, May 04, 2005 6:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IP Phones for home use? On 03/05/05, Justin B Newman [EMAIL PROTECTED] wrote: Neil Cherry wrote: What are your recommendations for a slightly fancy home phone? The Sipura SPA-841 is a nice compromise between the Ciscos and the Grandstreams. Check out the new Grandstream GXP-2000. I've been testing these, they're much better than the BT-100s. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN transfer, handoff to masterswitch
Hi Peter! Thanks for the quick response. So I'm already doing ECT by using the bristuff'ed version of *? Alex Mack Peter Svensson schrieb: On Wed, 4 May 2005, Alex Mack wrote: I've setup * with a Junghanns.net QuadBRI card agaisnt an Ericsson MD-110 PBX. All four ISDN channels are setup to simulate EuroISDN Point-to-Point (Anlagenanschluss in Germany) from the Ericsson's side. Works well and I have had little problems at all. Now what's happening during a call transfer of an ISDN call to another ISDN destination? Does such a transfer block 2 b-channels (one in and one out)? Or is it automatically handed off to the Ericsson so * is out of the loop? Is EuroISDN capable of such a handoff? Or do I have to use another protocol like Q.SIG for that feature? In EuroISDN this functionallity is provided by ECT (Explicit Call Transfer) for established calls and CD (Call Deflection) for calls in the setup phase. Bristuff supports these according to the documentation. Q.SIG has one (or possibly several) ways of doing this. At least one of these have been implemented in libpri. I don't have experience with Q.SIG at all. I was happy * worked with the EuroISDN trunk so well. But there's the issue with congestioning the link to the PBX with too many (in my case 4) forwarded calls. Does anyone have experiences with this? Are there references for Q-SIG out there? I think at least some of the standards are published at http://www.ecma-international.org/activities/Communications/QSIG_page.htm Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk CDR - Mysql
If cdr mysql status is 'command not found' then that means you haven't loaded the module. Check your module path to make sure it really is there. (/usr/lib/asterisk/modules/) If it is indeed there, do load cdr_addon_mysql.so from CLI* You might want to check modules.conf and make sure you have an autoload in there. -Matthew Rick Baranowski wrote: We seem to be having the same problem. The cdr command is not found, so we tried to do a make and install on the add-ons but it can't see to find the files when we run 'make clean make make install'. We have downloaded from CVS and the files look to be there but it still can't find the files. Could someone help? Thanks Rick [EMAIL PROTECTED] asterisk-addons]# make clean make make install rm -f *.so *.o .depend make -C format_mp3 clean make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' rm -f *.o *.so *~ make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` app_addon_sql_mysql.c:31:19: mysql.h: No such file or directory cdr_addon_mysql.c:33:19: mysql.h: No such file or directory cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o common.o common.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o dct64_i386.o dct64_i386.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o decode_ntom.o decode_ntom.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o layer3.o layer3.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o tabinit.o tabinit.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o interface.o interface.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o format_mp3.o format_mp3.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -shared -Xlinker -x -o format_mp3.so common.o dct64_i386.o decode_ntom.o layer3.o tabinit.o interface.o format_mp3.o make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' for x in format_mp3/format_mp3.so ; do install -m 755 $x /usr/lib/asterisk/modules ; done [EMAIL PROTECTED] asterisk-addons]# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Monday, May 02, 2005 7:03 PM To: Asterisk Users Subject: Re: [Asterisk-Users] Asterisk CDR - Mysql If you enabled it in logger.conf, it should be at /var/log/asterisk/debug What does cdr mysql status do? If it says no such command then you haven't loaded the cdr module. Did you do make install inside the asterisk-addons dir? Do you have autoload = yes in your modules.conf? -Matthew From: Callum McGillivray [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tue, 03 May 2005 12:03:17 +1000 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk CDR - Mysql Which Debug log ? Where is it located ? I can't see anything obvious that shows this info. Cheers, Callum (P.S. I'm not seeing a connection on the mySQL DB from the asterisk machine, and I assumed that there should be one... what am I missing here ? ) Matthew Boehm wrote: What is in your debug log? It will show the exact SQL that is being executed. -Matthew From: Callum McGillivray [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tue, 03 May 2005 11:35:52 +1000 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk CDR - Mysql Hi All, We have configured our Asterisk Server (CVS Head) to use mysql for CDR's, following the guidelines located at http://www.voip-info.org/wiki-Asterisk+cdr+mysql . When Asterisk starts up there are no errors, when we make a call there are no errors, however I am not seeing records in the database. Any idea how what I should be looking for here? I'm a bit lost. Cheers, Callum ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Post Paid Billing
Ezekiel Smith wrote: Could somebody recommend a good software utility, preferably with a web front, end for post paid billing in Asterisk? I've seen a lot of discussion on the various pre-paid and calling card based solutions, but nothing that would allow me to configure different regex-based locations/costs and generate a bill for a given user (users sorted out by SetAccount app and resulting application codes appended to CDR) at the END of the month. Thank you, EZ There is no turn key script for asterisk billing as every company is different. You're better off just writing one from scratch. We did. Calculates bills, creates PDF, emails customers, links to online CC payment. All in PHP. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Put a wait in a .call file.
Ronan Eckelberry wrote: Does anyone know of a way to put a wait or a pause in a .call file? When my * tries to make an outgoing call on a Zap channel, it does not wait for a dialtone. It just starts dialing. Thanks, -Ronan This seems to be a serious shortcoming in Asterisk. Can anyone explain why listening for dialtone wasn't an early consideration? With all the toneplans , by country, that are defined, it seems this was considered, but then never made to work John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mysql/Radius Authentication
Interestingly enough, your subject says Radius yet you didn't say anything about Radius in your email MySQL auth on 1.0.7 was removed (I think). It might still be there but it doesn't support NAT nor MWI. Just download CVS and use RealTime. We are using yesterdays CVS in a production environment and haven't had any problems. -Matthew Ganbold Tsagaankhuu wrote: Hi all, I'm using asterisk-1.0.7. I need to configure asterisk in such way that it authenticates users from mysql DB. Is it possible to authenticate SIP users from mysql database? It seems to me that chan_sip2 code from Olle E. Johansson, Edvina.net, [EMAIL PROTECTED] can authenticate users from mysql. However I looked for it everywhere and didn't find. Where can I download chan_sip2 code? Is there any other way I can authenticate SIP users from mysql in asterisk? Is it possible to make asterisk work with radius? I appreciate if somebody can give me some hints and advices in this regard. thanks in advance, Ganbold ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bad CLI colors? bad terminal?
Dave Cotton wrote: On Tue, 2005-05-03 at 20:05 -0500, Matthew Boehm wrote: Holy crap! You mean someone actually read my email? Thanks Andrew. Wish more people would read emails. Just read it :) I run from safe_asterisk and have the line ASTARGS=-n in it. Because I too hate the changing background. WFM(tm) I did this: [EMAIL PROTECTED] root]# ASTARGS=-n [EMAIL PROTECTED] root]# asterisk -Rvvdgn Then made a call. The black background shows up right at the G of the first goto statement. Would it make a difference if asterisk was started with -n or not? Seems to me that even if asterisk was started without -n that asterisk should honor any subsequent reattachments CLI options. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Philips - [QSIG] - Alcatel - [H323] - Asterisk - [SIP] - Users.
Hi everybody, Firstly we have to connect our Asterisk system to a Philips PBX throught QSIG protocol (interfaces S0), but we doesn't find any documentation about the support of QSIG and S0 interfaces by Asterisk. [PSTN/ISDN] --- Philips -[QSIG over S0]- Asterisk -[SIP]- Final users. Is it possible? does Asterisk support QSIG and S0 interfaces? Thinking a bit more, we have defined thissecond scenario: [PSTN/ISDN] --- Philips -[QSIG over S0]- Alcatel -[H.323]- Asterisk -[SIP]- Final users. Do Asterisk functions of H323/SIP Gateway? Have to install any aditional software to Asterisk to do Gateway functions? Any suggested scenario to do this integration? Regards. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Put a wait in a .call file.
Put the call file into a folder and have cron copy it to the outgoing spool after a pause Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronan Eckelberry Sent: Wednesday, May 04, 2005 9:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Put a wait in a .call file. Does anyone know of a way to put a wait or a pause in a .call file? When my * tries to make an outgoing call on a Zap channel, it does not wait for a dialtone. It just starts dialing. Thanks, -Ronan NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bad CLI colors? bad terminal?
On Wed, 2005-05-04 at 08:46 -0500, Matthew Boehm wrote: Dave Cotton wrote: On Tue, 2005-05-03 at 20:05 -0500, Matthew Boehm wrote: Holy crap! You mean someone actually read my email? Thanks Andrew. Wish more people would read emails. Just read it :) I run from safe_asterisk and have the line ASTARGS=-n in it. Because I too hate the changing background. WFM(tm) I did this: [EMAIL PROTECTED] root]# ASTARGS=-n [EMAIL PROTECTED] root]# asterisk -Rvvdgn Then made a call. The black background shows up right at the G of the first goto statement. Would it make a difference if asterisk was started with -n or not? Seems to me that even if asterisk was started without -n that asterisk should honor any subsequent reattachments CLI options. Hey, I said I _read_ yours :) The ASTARGS=-n is in the script safe_asterisk. It of course means you have to do a service stop/start to get it to kick in but after that ça marche. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Put a wait in a .call file.
John, Since you think its a serious shortcoming, either you fix it or you shut up. To start bitching here and complain that its considered and not implemented is bullshit. * is a great product, but all great product has their flaws. Being OSS, you can always modify the code yourself. Otherwise just ask nicely and someone probably wouldn't mind helping. Best Regards, David Choo John Novack [EMAIL PROTECTED] g-carlson.org To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 04/05/2005 09:48 Re: [Asterisk-Users] Put a wait in PMa .call file. Please respond to [EMAIL PROTECTED] -carlson.org; Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com Ronan Eckelberry wrote: Does anyone know of a way to put a wait or a pause in a .call file? When my * tries to make an outgoing call on a Zap channel, it does not wait for a dialtone. It just starts dialing. Thanks, -Ronan This seems to be a serious shortcoming in Asterisk. Can anyone explain why listening for dialtone wasn't an early consideration? With all the toneplans , by country, that are defined, it seems this was considered, but then never made to work John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SetCallerPres problem
Hi, Background: I'm running 2x * boxes. Box A has a registered user which dials a number. The connection is sent to Box B which acts as pstn gateway (sangoma 1xE1 card). Problem: On Box A before executing Dial() command I set SetCallerPres(prohib_no_screened) but despite that Box B sends the connection to pstn with allowed_not_screened flag ? Why is that? When I set SetCallerPres(prohib_no_screened) on Box B it acts properly. But why sending this flag between 2 8 boxes doesn't work for me? Any suggestions? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] monitoring which IVR extension is pressed
Is there anyway of monitoring which extension is pressed on a IVR, I need to use it for voting application. Look at AGI (or system() if you already have scripts) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bad CLI colors? bad terminal?
Dave Cotton wrote: On Wed, 2005-05-04 at 08:46 -0500, Matthew Boehm wrote: Dave Cotton wrote: On Tue, 2005-05-03 at 20:05 -0500, Matthew Boehm wrote: Holy crap! You mean someone actually read my email? Thanks Andrew. Wish more people would read emails. Just read it :) I run from safe_asterisk and have the line ASTARGS=-n in it. Because I too hate the changing background. WFM(tm) I did this: [EMAIL PROTECTED] root]# ASTARGS=-n [EMAIL PROTECTED] root]# asterisk -Rvvdgn Then made a call. The black background shows up right at the G of the first goto statement. Would it make a difference if asterisk was started with -n or not? Seems to me that even if asterisk was started without -n that asterisk should honor any subsequent reattachments CLI options. Hey, I said I _read_ yours :) The ASTARGS=-n is in the script safe_asterisk. It of course means you have to do a service stop/start to get it to kick in but after that ça marche. Why would I have to start the server without colors? Why can't I just reconnect to the server and during that single instance, not have colors? Seems very microsoft-ish to force the client to use the same settings as the server. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ackcall
Is there a way to have an agent choose whether they want to press # to accept a call on an individual basis when they log in? Also, the faq mentions that you can play an optional message to the agent before they press '#', how is this performed? The queue message seems to play AFTER they press the '#'. Thanks, Jon. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Put a wait in a .call file.
or create a file in another dir. Change the time on the file then put it in the call spool. It should be covered on the WIKI as well. Or you could write your own app to use the manager api to originate the calls depending on the needs you have. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Philips - [QSIG] - Alcatel - [H323] - Asterisk -[SIP] - Users.
DT wrote: Firstly we have to connect our Asterisk system to a Philips PBX throught QSIG protocol (interfaces S0), but we doesn't find any documentation about the support of QSIG and S0 interfaces by Asterisk. [PSTN/ISDN] --- Philips -[QSIG over S0]- Asterisk -[SIP]- Final users. Is it possible? does Asterisk support QSIG and S0 interfaces? As far as I know, Asterisk doesn't support QSIG. Do you _have to_ use QSIG? I'd just use a PRI interface (DTU-PH IIRC) to connect to Asterisk with a sutable PCI card in the server. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Put a wait in a .call file.
Add some 'w' before the number, i.e., Zap/g0/ww1812121212 Julian J. M. On 5/4/05, Ronan Eckelberry [EMAIL PROTECTED] wrote: Does anyone know of a way to put a wait or a pause in a .call file? When my * tries to make an outgoing call on a Zap channel, it does not wait for a dialtone. It just starts dialing. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Put a wait in a .call file.
Another social misfit appears. Or is it a full moon tonight? Didn't your mother teach you any manners? John Novack David Choo wrote: John, Since you think its a serious shortcoming, either you fix it or you shut up. To start bitching here and complain that its considered and not implemented is bullshit. * is a great product, but all great product has their flaws. Being OSS, you can always modify the code yourself. Otherwise just ask nicely and someone probably wouldn't mind helping. Best Regards, David Choo John Novack [EMAIL PROTECTED] g-carlson.org To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 04/05/2005 09:48 Re: [Asterisk-Users] Put a wait in PMa .call file. Please respond to [EMAIL PROTECTED] -carlson.org; Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com Ronan Eckelberry wrote: Does anyone know of a way to put a wait or a pause in a .call file? When my * tries to make an outgoing call on a Zap channel, it does not wait for a dialtone. It just starts dialing. Thanks, -Ronan This seems to be a serious shortcoming in Asterisk. Can anyone explain why listening for dialtone wasn't an early consideration? With all the toneplans , by country, that are defined, it seems this was considered, but then never made to work John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM04B in a Mac
As anyone been able to make a TDM04B work in a Mac with Yellow Dog 3.01? (unless I have to use another version of Yellow Dog?) I tried on a Power Mac 8500, a G3 Beige Desktop, G3 Blue White and G4 tower... I can compile zaptel and asterisk witthout any problem. The card is seen but when I try to make a call or when I receive one I have no audio at all on aany of the computer above... I know the card is working fine in a PC I tested it with the same config. I heard people been able to make an X100P card work in a Mac but couldn't find anything about a TDM400 card... Anyone found a way to make it work? Thanks Martin Roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 'multi-line' configuration
Pat, To my knowledge the only way to turn on and off the Call Waiting function is on-screen with the phone itself. There are quite a few of these 'little' features I wish would be configurable via the config file but don't seem to be... Best wishes, -Corey Great info! The only question I would have is on the call waiting setting. What should it be set to, and is the setting the one in the SIPX.conf file? Pat -- Corey S. McFadden ([EMAIL PROTECTED]) McFadden Associates - Technology Consultants phone 215-825-2121 x510 - web.csma.biz * This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ackcall
Jon Gabrielson wrote: Is there a way to have an agent choose whether they want to press # to accept a call on an individual basis when they log in? Also, the faq mentions that you can play an optional message to the agent before they press '#', how is this performed? The queue message seems to play AFTER they press the '#'. Thanks, Jon. Go read the wiki. You answer is there. Actually, if you read the sample config..you answer is also there. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Philips - [QSIG] - Alcatel - [H323] - Asterisk-[SIP] - Users.
Yes, we have to use QSIG directly with Asterisk or througt the Alcatel. So, Asterisk will be able to handle H.323 to redirect to correct SIP users? Regards. - Original Message - From: Andreas Sikkema [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 04, 2005 4:26 PM Subject: RE: [Asterisk-Users] Philips - [QSIG] - Alcatel - [H323] - Asterisk-[SIP] - Users. DT wrote: Firstly we have to connect our Asterisk system to a Philips PBX throught QSIG protocol (interfaces S0), but we doesn't find any documentation about the support of QSIG and S0 interfaces by Asterisk. [PSTN/ISDN] --- Philips -[QSIG over S0]- Asterisk -[SIP]- Final users. Is it possible? does Asterisk support QSIG and S0 interfaces? As far as I know, Asterisk doesn't support QSIG. Do you _have to_ use QSIG? I'd just use a PRI interface (DTU-PH IIRC) to connect to Asterisk with a sutable PCI card in the server. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newer Dell Servers + TDM card
Is this with the TDM400P card right? -Original Message- From: David Brodbeck [mailto:[EMAIL PROTECTED] Sent: Monday, May 02, 2005 2:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Newer Dell Servers + TDM card -Original Message- From: Matt Schulte [mailto:[EMAIL PROTECTED] Really, how long does it take to recover? Mine just totally locks. No time at all. The only reason I know an NMI occurs is the front panel light, and the Dazed and confused, but trying to continue message from the kernel. I'm using a Dell PowerEdge 800. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newer Dell Servers + TDM card
We have this issue with every new Dell server out, we've tried different distros even. A poweredge 800 was the last one we tried it on. It just locks hard, don't get it. We're using the TDM400P (Not T1).. -Original Message- From: David John Walsh [mailto:[EMAIL PROTECTED] Sent: Monday, May 02, 2005 3:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Newer Dell Servers + TDM card Matt Which server is this an issue with, I am looking to get power edge rack 1850's (the 1U) or 2850's the 2U? and which card are you refering to (I assume its a TE405p) thanks David On 5/2/05, Matt Schulte [EMAIL PROTECTED] wrote: Really, how long does it take to recover? Mine just totally locks. Matt -Original Message- From: David Brodbeck [mailto:[EMAIL PROTECTED] Sent: Monday, May 02, 2005 12:10 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Newer Dell Servers + TDM card -Original Message- From: Matt Schulte [mailto:[EMAIL PROTECTED] Has anyone ever been able to fix this NMI power issue that the Dell's have with the TDM cards? Basically locks the machine up when trying to bring up the module. I get an NMI the first time I load the module, but the machine always recovers. Subsequent load/unload cycles don't trigger further NMIs. I'd like to know of any way to fix it, too, 'cause that orange flashing light is kind of annoying. ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to display info from Asterisk on/to the phone ?
Hello, Thank you for replying ! These files are on the cisco ? or with Asterisk ? Because I don't have Cisco phone. Is there a way independant of the phone ? Best regards Le mardi 03 mai 2005 19:22 -0500, Ing CIP Alejandro Celi Maritegui a crit : El mar, 03-05-2005 a las 03:43, Deborah MALKA escribi: Hello, I wanted to know if there is a way to dissplay infos from Asterisk on a SIP phone ? Because I know Asterisk is very powerfull, so I'm nearly sure that there is a way to do it. Using XML on Directory.xml and services.xml with a Cisco 7960/7940 phone. I combine it with PHP Regqrds, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM04B in a Mac
The best way to get PSTN into a Mac or Windows Asterisk setup is with a Sipura SPA-3000. You can set it up as a trunk and it works great. I am actually working on an article on how to configure it right now. Kerry http://geekgazette.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Roy Sent: Wednesday, May 04, 2005 7:37 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TDM04B in a Mac As anyone been able to make a TDM04B work in a Mac with Yellow Dog 3.01? (unless I have to use another version of Yellow Dog?) I tried on a Power Mac 8500, a G3 Beige Desktop, G3 Blue White and G4 tower... I can compile zaptel and asterisk witthout any problem. The card is seen but when I try to make a call or when I receive one I have no audio at all on aany of the computer above... I know the card is working fine in a PC I tested it with the same config. I heard people been able to make an X100P card work in a Mac but couldn't find anything about a TDM400 card... Anyone found a way to make it work? Thanks Martin Roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Problems with TDM400P card
On Tue, May 03, 2005 at 05:27:33PM +, Tony Mountifield wrote: In article [EMAIL PROTECTED], Rich Adamson [EMAIL PROTECTED] wrote: - a modified zttest.c run on both systems to show the delays in reading 8192 bytes from the TDM card as 23,850 microseconds lateness on the old mobo, and 24,000 microsecond lateness on the new system. No significant change resulting from the differences in mobo, pci structure, interrupt structure, cpu speed, quantity of ram, kernel differences (v2.4 vs v2.6), etc. See my response to your zttest-mod.c posting. I think it is 8000 bytes that are due every second, not 8192. That would make the timing on your new system pretty accurate if 8192 bytes are arriving in 1,024,000us. Tony is correct. You should expect 8000 octets/sec from a digital sampler on a POTS line interface. http://lists.digium.com/pipermail/asterisk-users/2005-May/105148.html Reference: http://www.ncta.com/industry_overview/cableGlossary.cfm?indOverviewID=41 Sample Rate - In analog to digital signal processing, the sample rate is the interval at which samples of an analog signal are taken. The sample rate for digital telephony, for example, is 8000 per second. http://www.freesoft.org/CIE/Topics/127.htm G.711 is the international standard for encoding telephone audio on an 64 kbps channel. It is a pulse code modulation (PCM) scheme operating at a 8 kHz sample rate, with 8 bits per sample. According to the Nyquist theorem, which states that a signal must be sampled at twice its highest frequency component, G.711 can encode frequencies between 0 and 4 kHz. Telcos can select between two different varients of G.711: A-law and mu-law. A-law is the standard for international circuits. -- Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk CDR - Mysql
Matthew, Thank you very much for the help. We know that the module is not loading because we can't do the make and make install successfully for the add-ons. It's telling us that it can't find the files necessary when we do a make(print out listed below). We have renamed the add-ons dir and downloaded again from the CVS but we are still getting this error. Any thoughts? Thanks Rick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Wednesday, May 04, 2005 6:38 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk CDR - Mysql If cdr mysql status is 'command not found' then that means you haven't loaded the module. Check your module path to make sure it really is there. (/usr/lib/asterisk/modules/) If it is indeed there, do load cdr_addon_mysql.so from CLI* You might want to check modules.conf and make sure you have an autoload in there. -Matthew Rick Baranowski wrote: We seem to be having the same problem. The cdr command is not found, so we tried to do a make and install on the add-ons but it can't see to find the files when we run 'make clean make make install'. We have downloaded from CVS and the files look to be there but it still can't find the files. Could someone help? Thanks Rick [EMAIL PROTECTED] asterisk-addons]# make clean make make install rm -f *.so *.o .depend make -C format_mp3 clean make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' rm -f *.o *.so *~ make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` app_addon_sql_mysql.c:31:19: mysql.h: No such file or directory cdr_addon_mysql.c:33:19: mysql.h: No such file or directory cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o common.o common.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o dct64_i386.o dct64_i386.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o decode_ntom.o decode_ntom.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o layer3.o layer3.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o tabinit.o tabinit.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o interface.o interface.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o format_mp3.o format_mp3.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -shared -Xlinker -x -o format_mp3.so common.o dct64_i386.o decode_ntom.o layer3.o tabinit.o interface.o format_mp3.o make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' for x in format_mp3/format_mp3.so ; do install -m 755 $x /usr/lib/asterisk/modules ; done [EMAIL PROTECTED] asterisk-addons]# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Monday, May 02, 2005 7:03 PM To: Asterisk Users Subject: Re: [Asterisk-Users] Asterisk CDR - Mysql If you enabled it in logger.conf, it should be at /var/log/asterisk/debug What does cdr mysql status do? If it says no such command then you haven't loaded the cdr module. Did you do make install inside the asterisk-addons dir? Do you have autoload = yes in your modules.conf? -Matthew From: Callum McGillivray [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tue, 03 May 2005 12:03:17 +1000 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk CDR - Mysql Which Debug log ? Where is it located ? I can't see anything obvious that shows this info. Cheers, Callum (P.S. I'm not seeing a connection on the mySQL DB from the asterisk machine, and I assumed that there should be one... what am I missing here ? ) Matthew Boehm wrote: What is in your debug log? It will show the exact SQL that is being executed. -Matthew From: Callum McGillivray [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tue, 03 May 2005 11:35:52 +1000 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk CDR - Mysql Hi All, We
Re: [Asterisk-Users] 7960 'multi-line' configuration
Corey S. McFadden wrote: Pat, To my knowledge the only way to turn on and off the Call Waiting function is on-screen with the phone itself. There are quite a few of these 'little' features I wish would be configurable via the config file but don't seem to be... # Call Waiting (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control) call_waiting: 2 ; Default 1 (Enable Call Waiting) ... a bunch of options are only listed if you browse through all the info on cisco.com. Regardless, there is the option to add to SIPDefault.cnf to make the phones do what you want in regards to Call Waiting, etc... -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] new production server for SOHO installation
I've just ordered up a new PC for my home office. I've decided upon a VIA 500 MHz platform in a fanless/silent case with one PCI slot for my TDM400 card. Instead of a HD I'm using an IDE CF convertor and AstLinux. To the user community I pose a question about throughput expectation on such a platform. If I someday decide to use G.729a where I now use G.711 to make outgoing calls through ITSPs what sort of limit can I expect? How many calls before I run out of CPU power? Also, does the onboard crypto engine (known as Padlock) in newer VIA chips have any potential to impact Asterisk? My understanding is that it allows the slower VIA chips to seriously speed up AES encryption. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cellsocket NEED HELP
I just got a cellsocket for my * box...I need help, will give you a channel on my box in exchange for your time to help me out, if you have experience to configure this things, please contact me out list... Manny Mawise(at)hotmaildotcom Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newer Dell Servers + TDM card
We have this issue with every new Dell server out, we've tried different distros even. A poweredge 800 was the last one we tried it on. It just locks hard, don't get it. We're using the TDM400P (Not T1).. For everyone's information, we are successfully using a TDM400P card with a single FXO module in a Dell Poweredge 2550 running Slackware 10. I know you can't get those servers any more, but they do make modern day equivalents in the form of 2850s. The 2650 was the successor to the 2550, and the precursor to the 2850, but I've not tested either of these later models. Thanks Stuart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channels ???
Can I send an receive call on the same channel (line to the wall) Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Collect calls
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 James, jltaylor escreveu: | Since you are referring to R2 signaling, it works like this: I'm referring to ISDN PRI channels not R2. | | The E1 R2 Call Blocking feature provides two ways to block incoming collect | calls-category-based and double answer. With category-based call blocking, | collect calls will be blocked based on a specific category. For example, in | Brazil, collect calls arrive with a category II-8, for which the gateway | should send B-7 as a response instead of an answer signal. This approach is | only applicable when switches in the central office support category-based | blocking. | | For legacy switches that do not support category-based blocking, the double | answer method is implemented to support the collect-call blocking. For an | incoming collect call, the gateway will answer the call with a clearback | after one second and re-answer the call after two seconds, causing the | collect call to be dropped and normal calls to stay connected. Can you give me an example using this method with Asterisk? | | This is what the referenced patches are attempting to do. Referenced Patches? What do you mean? Does someone is working with patches to implement this feature in Asterisk? | | This does not work in the U.S. or if you have SS7, you don't need it. Thanks for your answer. | | James | | -Original Message- | From: [EMAIL PROTECTED] | [mailto:[EMAIL PROTECTED] Behalf Of Michael D | Schelin | Sent: Tuesday, May 03, 2005 6:06 PM | To: Asterisk Users Mailing List - Non-Commercial Discussion | Subject: Re: [Asterisk-Users] Collect calls | | | You Bring up a great point. I understand these codes and my system | brings them in via ss7 but as youself I don't know how to protect my | network from these charges. I will follow this post to see if anybody | has a fix. | | | Rodrigo P. Telles wrote: | | | Hi Folks, | | Does someone knows how to identify and block collect calls on Asterisk | using PRI | channels? | I googled it and found this: | http://lists.digium.com/pipermail/asterisk-dev/2004-November/007500.html | I don't know what does it mean!!! | Can someone help me to understand this? | | I tried to apply that way too, using Flash() but Flash() complains and | looks | like just work with FXO channels: | http://lists.digium.com/pipermail/asterisk-users/2004-October/066360.html | | Thanks in advance. | | -- | | Rodrigo P. Telles [EMAIL PROTECTED] | IVOZ # 1009 | TI Manager | Devel-IT - http://www.devel.it | Bestcom Group | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: ~ http://lists.digium.com/mailman/listinfo/asterisk-users | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users - -- Rodrigo P. Telles [EMAIL PROTECTED] IVOZ # 1009 TI Manager Devel-IT - http://www.devel.it Bestcom Group -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFCeOf0iLK8unYgEMQRAnFJAJoDdR07uKNGOyIjtV1lgnrCoS+7xACfTRc/ aaw9DBci1lZfamMxO4PQJdA= =Y/Qc -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: TDM users: modified zttest.c for testing
On Tue, May 03, 2005 at 09:28:23PM +, Tony Mountifield wrote: I wrote: In article [EMAIL PROTECTED], Rich Adamson [EMAIL PROTECTED] wrote: It would be very interesting to see everyone's results in running this, and even more interesting to report the results with the OS distro in use, mobo in use (if known), etc. If anyone actually get's a result that is very close to 1.000 seconds, I'd really like to know more about those systems. (email off list is fine if you want.) --- Results after 20 passes --- Best: 1.024003 -- Worst: 1.023981 -- Average: 1.023993 This looks very close to 1024ms instead of 1000ms. That got me thinking: I believe your premise is wrong. The sample rate of telephony audio is 8kHz. With 8-bit samples (uLaw or aLaw), that means 8000 bytes should be supplied in 1 second, not 8192. At a rate of 8000 bytes/sec, 8192 bytes will arrive in 1.024 seconds. snip [EMAIL PROTECTED] zaptel]# ./zttest-mod -v Objective: to read 8000 bytes from TDM card in 1.00 seconds. Opened pseudo zap interface, measuring accuracy... read(fd, buf, 8000) returns 1024 read(fd, buf, 6976) returns 1024 read(fd, buf, 5952) returns 1024 read(fd, buf, 4928) returns 1024 read(fd, buf, 3904) returns 1024 read(fd, buf, 2880) returns 1024 read(fd, buf, 1856) returns 1024 read(fd, buf, 832) returns 832 Whew! At least the kernel module is using the len :). 8000 bytes in 1.023988 seconds read(fd, buf, 8000) returns 1024 read(fd, buf, 6976) returns 1024 read(fd, buf, 5952) returns 1024 read(fd, buf, 4928) returns 1024 read(fd, buf, 3904) returns 1024 read(fd, buf, 2880) returns 1024 read(fd, buf, 1856) returns 1024 read(fd, buf, 832) returns 832 8000 bytes in 1.023998 seconds --- Results after 2 passes --- Best: 1.023998 -- Worst: 1.023988 -- Average: 1.023993 [EMAIL PROTECTED] zaptel]# So it looks like the pseudo driver is always handling 1024 byte chunks, and even if you ask it for fewer bytes, it takes 1024 bytes' worth of time. I think it should really be handling 1000-byte chunks in 125ms rather than 1024-byte chunks in 128ms, if it is supposed to be emulating telephony channels. Why? Computers are base-2 oriented and POTS digital telephony is based on adapting to human hearing perception and a massive installed base of analog equipment. Working with base-2 numbers in computer programs is common and often efficient. 1/8000 = 0.000125 sec/sample octet (8 * 1024)samples * 0.000125 sec/sample = 1.024 sec Looks good to me. But zaptel.c is Deep Magic, and I'd be interested in comments from those who are famliar with it in detail. Bah. Just your average 6459 line kernel module ;). I've seen bigger. Here are some guides: http://www.oreilly.com/catalog/linuxdrive2/ http://kernelnewbies.org/documents/kdoc/kernel-api/linuxkernelapi.html -- Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk CDR - Mysql
From your make output it looks like maybe you don't have the mySQL development package installed on this box? The one with the associated header files, etc. On the RPM-based Linux systems I've used (like Redhat, Mandrake, CentOS, Suse) the package is usually named [packagname]-devel.version.rpm, so you might search your install CD's for a mysql-devel-#.#.#.#.rpm file and try installing that (rpm -ivh [pacakgename]), then try compiling again. Brent -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rick Baranowski Sent: Wednesday, May 04, 2005 10:14 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk CDR - Mysql Matthew, Thank you very much for the help. We know that the module is not loading because we can't do the make and make install successfully for the add-ons. It's telling us that it can't find the files necessary when we do a make(print out listed below). We have renamed the add-ons dir and downloaded again from the CVS but we are still getting this error. Any thoughts? Thanks Rick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Wednesday, May 04, 2005 6:38 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk CDR - Mysql If cdr mysql status is 'command not found' then that means you haven't loaded the module. Check your module path to make sure it really is there. (/usr/lib/asterisk/modules/) If it is indeed there, do load cdr_addon_mysql.so from CLI* You might want to check modules.conf and make sure you have an autoload in there. -Matthew Rick Baranowski wrote: We seem to be having the same problem. The cdr command is not found, so we tried to do a make and install on the add-ons but it can't see to find the files when we run 'make clean make make install'. We have downloaded from CVS and the files look to be there but it still can't find the files. Could someone help? Thanks Rick [EMAIL PROTECTED] asterisk-addons]# make clean make make install rm -f *.so *.o .depend make -C format_mp3 clean make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' rm -f *.o *.so *~ make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` app_addon_sql_mysql.c:31:19: mysql.h: No such file or directory cdr_addon_mysql.c:33:19: mysql.h: No such file or directory cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o common.o common.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o dct64_i386.o dct64_i386.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o decode_ntom.o decode_ntom.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o layer3.o layer3.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o tabinit.o tabinit.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o interface.o interface.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o format_mp3.o format_mp3.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -shared -Xlinker -x -o format_mp3.so common.o dct64_i386.o decode_ntom.o layer3.o tabinit.o interface.o format_mp3.o make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' for x in format_mp3/format_mp3.so ; do install -m 755 $x /usr/lib/asterisk/modules ; done [EMAIL PROTECTED] asterisk-addons]# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Monday, May 02, 2005 7:03 PM To: Asterisk Users Subject: Re: [Asterisk-Users] Asterisk CDR - Mysql If you enabled it in logger.conf, it should be at /var/log/asterisk/debug What does cdr mysql status do? If it says no such command then you haven't loaded the cdr module. Did you do make install inside the asterisk-addons dir? Do you have autoload = yes in your modules.conf? -Matthew From: Callum McGillivray [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tue, 03 May 2005 12:03:17 +1000 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject:
[Asterisk-Users] Cisco 7960: Builtin CFwdAll working?
Hey guys, Lots of nice people on the list using 7960s and using/discovering features that I didn't think possible. (Re: Multi Line Appearance). Wanted to know if anyone has gotten the 'CFwdAll' button to properly work. The problem I am seeing is that if someone presses the button and types in their cell (for instance), I get a local channel into a loop message in asterisk. Here is an incomming call to my DID to my 7960 (x3044) -- Called 3044 -- Got SIP response 302 Moved Temporarily back from 10.0.0.36 -- Now forwarding Zap/1-1 to 'SIP/[EMAIL PROTECTED]:5060' (thanks to SIP/3044-a649) -- Got SIP response 482 Loop Detected back from 10.0.3.10 -- Now forwarding Zap/1-1 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/10.0.3.10:5060-a32b) May 4 10:28:21 NOTICE[25650]: chan_local.c:436 local_alloc: No such extension/context [EMAIL PROTECTED] creating local channel May 4 10:28:21 NOTICE[25650]: app_dial.c:355 wait_for_answer: Unable to create local channel for call forward to 'Local/[EMAIL PROTECTED]' (cause = 0) The problem is, that the phone makes a brand new SIP call to asterisk. Well, all incomming calls go into the all-incomming context. I was expecting the new call to use the same context that the phone is registered into. Any thoughts? Ideas? Thanks, Matthew -- Matthew Boehm, IT DirectorCypress Telecommunications [EMAIL PROTECTED] 3838 N. Sam Houston Parkway E #400 T: 832-200-8640 x3044 Houston, TX 77032 My girlfriend was recently diagnosed with multiple personality disorder; When she called yesterday, my CallerID box exploded. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk CDR - Mysql
Doh. I didn't read close enough. app_addon_sql_mysql.c:31:19: mysql.h: No such file or directory cdr_addon_mysql.c:33:19: mysql.h: No such file or directory cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory That means you don't have mysql installed. Or rather, you don't have mysql headers/libraries stored in default locations. mysql.h should be located (by default) in /usr/local/include/mysql/ other common locations: /usr/include/ /usr/include/mysql/ -Matthew Rick Baranowski wrote: Matthew, Thank you very much for the help. We know that the module is not loading because we can't do the make and make install successfully for the add-ons. It's telling us that it can't find the files necessary when we do a make(print out listed below). We have renamed the add-ons dir and downloaded again from the CVS but we are still getting this error. Any thoughts? Thanks Rick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Wednesday, May 04, 2005 6:38 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk CDR - Mysql If cdr mysql status is 'command not found' then that means you haven't loaded the module. Check your module path to make sure it really is there. (/usr/lib/asterisk/modules/) If it is indeed there, do load cdr_addon_mysql.so from CLI* You might want to check modules.conf and make sure you have an autoload in there. -Matthew Rick Baranowski wrote: We seem to be having the same problem. The cdr command is not found, so we tried to do a make and install on the add-ons but it can't see to find the files when we run 'make clean make make install'. We have downloaded from CVS and the files look to be there but it still can't find the files. Could someone help? Thanks Rick [EMAIL PROTECTED] asterisk-addons]# make clean make make install rm -f *.so *.o .depend make -C format_mp3 clean make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' rm -f *.o *.so *~ make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` app_addon_sql_mysql.c:31:19: mysql.h: No such file or directory cdr_addon_mysql.c:33:19: mysql.h: No such file or directory cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o common.o common.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o dct64_i386.o dct64_i386.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o decode_ntom.o decode_ntom.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o layer3.o layer3.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o tabinit.o tabinit.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o interface.o interface.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o format_mp3.o format_mp3.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -shared -Xlinker -x -o format_mp3.so common.o dct64_i386.o decode_ntom.o layer3.o tabinit.o interface.o format_mp3.o make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' for x in format_mp3/format_mp3.so ; do install -m 755 $x /usr/lib/asterisk/modules ; done [EMAIL PROTECTED] asterisk-addons]# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Monday, May 02, 2005 7:03 PM To: Asterisk Users Subject: Re: [Asterisk-Users] Asterisk CDR - Mysql If you enabled it in logger.conf, it should be at /var/log/asterisk/debug What does cdr mysql status do? If it says no such command then you haven't loaded the cdr module. Did you do make install inside the asterisk-addons dir? Do you have autoload = yes in your modules.conf? -Matthew From: Callum McGillivray [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tue, 03 May 2005 12:03:17 +1000 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk CDR - Mysql Which Debug log ? Where is it located ? I can't see anything obvious that shows this info. Cheers, Callum (P.S. I'm not
[Asterisk-Users] HFC: zapata + bristuff - how to set an outgoing number
I have a HFC-PCI based ISDN card. How should an extension be constructed, when I want to set up a specific outgoing number (I have 10 or so MSN numbers)? For example, when I call 6546 from my SIP phone, I would like to call 100 with an outgoing number of 555 - how should I do this? exten = 5646,1,Dial(Zap/g0/98) Tomek -- Startuj z INTERIA.PL! http://link.interia.pl/f186c ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI timing problems: Fax Voice
I've been trying to get PRI - Email Fax to work for some time now. Several months. Got newest everything and still some pages come out missing an inch or two. It was recommended to me to change my zaptel.conf so that span #1 used itself as primary sync source. (It was set to 0). I made the change and now FAXES LOOK PERFECT!!! 100+ pages and not a single problem. The problem now is, I get this error every so often: May 4 10:57:04 WARNING[25650]: chan_zap.c:4394 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 2 - audio may have been lost When the span was 0, I NEVER got that message. I haven't heard any complaints from the other office mates that use the PRI for voice, but the error just bothers me. What is the real difference between 0 and 1 on the span timing? Thanks, Matthew -- Matthew Boehm, IT DirectorCypress Telecommunications [EMAIL PROTECTED] 3838 N. Sam Houston Parkway E #400 T: 832-200-8640 x3044 Houston, TX 77032 My girlfriend was recently diagnosed with multiple personality disorder; When she called yesterday, my CallerID box exploded. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM users: modified zttest.c for testing
On Wed, 2005-05-04 at 06:48 -0600, Rich Adamson wrote: On Tue, 2005-05-03 at 09:48 -0600, Rich Adamson wrote: TDM X100P card users: I get average numbers very close to 1.024 (especially if I take some rounding error into account). That's a very good point. Now I'm not sure since the only thing I've got to go by is existing code in zttest.c which implies 8192, and data arrives in 1024 byte frames. I'll dig a little deeper to see if I can figure out which one _is_ correct. I also recommend only printing 3 decimal places for the times. All of the additional digits are just noise. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960: Builtin CFwdAll working?
It works for me. Do you have reinvites enabled. I do not. That may explain why * is sending a redirect. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Wednesday, May 04, 2005 11:35 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960: Builtin CFwdAll working? Hey guys, Lots of nice people on the list using 7960s and using/discovering features that I didn't think possible. (Re: Multi Line Appearance). Wanted to know if anyone has gotten the 'CFwdAll' button to properly work. The problem I am seeing is that if someone presses the button and types in their cell (for instance), I get a local channel into a loop message in asterisk. Here is an incomming call to my DID to my 7960 (x3044) -- Called 3044 -- Got SIP response 302 Moved Temporarily back from 10.0.0.36 -- Now forwarding Zap/1-1 to 'SIP/[EMAIL PROTECTED]:5060' (thanks to SIP/3044-a649) -- Got SIP response 482 Loop Detected back from 10.0.3.10 -- Now forwarding Zap/1-1 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/10.0.3.10:5060-a32b) May 4 10:28:21 NOTICE[25650]: chan_local.c:436 local_alloc: No such extension/context [EMAIL PROTECTED] creating local channel May 4 10:28:21 NOTICE[25650]: app_dial.c:355 wait_for_answer: Unable to create local channel for call forward to 'Local/[EMAIL PROTECTED]' (cause = 0) The problem is, that the phone makes a brand new SIP call to asterisk. Well, all incomming calls go into the all-incomming context. I was expecting the new call to use the same context that the phone is registered into. Any thoughts? Ideas? Thanks, Matthew -- Matthew Boehm, IT DirectorCypress Telecommunications [EMAIL PROTECTED] 3838 N. Sam Houston Parkway E #400 T: 832-200-8640 x3044 Houston, TX 77032 My girlfriend was recently diagnosed with multiple personality disorder; When she called yesterday, my CallerID box exploded. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Attended Transfer using wrong Context
The phone's context is cytel-internal. This allows us to hit 3XXX to get someone on the inside. If you hit 9 at the beginning, you Goto() the cytel-outgoing context. So lets make a call..I'll dial 918005551212 (toll free directory). The 9 sends it to cytel-outgoing. Call is made. Bridged. I then hit #9 for attended transfer. Allison says Transfer. I start to enter 3013. But right after I hit the first 3, it returns failed transfer: res_features.c:800 builtin_atxfer: Did not read data. Wtf? So I do it again; and again. I tried every number and they all returned the same error. But this time I press 93013 and the call goes out the cytel-outgoing context. ???!?? I'm lost. What is this thing doing? -Matthew -- Matthew Boehm, IT DirectorCypress Telecommunications [EMAIL PROTECTED] 3838 N. Sam Houston Parkway E #400 T: 832-200-8640 x3044 Houston, TX 77032 My girlfriend was recently diagnosed with multiple personality disorder; When she called yesterday, my CallerID box exploded. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HFC: zapata + bristuff - how to set an outgoing number
exten = 5646,1,SetCallerID(some name 555) exten = 5646,2,Dial(Zap/g0/98) http://voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetCallerID On Wed, 2005-05-04 at 08:53, Tomasz Chmielewski wrote: I have a HFC-PCI based ISDN card. How should an extension be constructed, when I want to set up a specific outgoing number (I have 10 or so MSN numbers)? For example, when I call 6546 from my SIP phone, I would like to call 100 with an outgoing number of 555 - how should I do this? exten = 5646,1,Dial(Zap/g0/98) Tomek -- Startuj z INTERIA.PL! http://link.interia.pl/f186c ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mac OS X proves that it's easier to make UNIX pretty than it is to make Windows secure signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queues configuration
We ran into the same problem. Found out by reading the source that we had to use joinempty=strict and leavewhenempty=strict to make it work. Now if I could just get it to pause the agent when someone direct dials the extension, and then unpause consistently when they hang up. Anton Krall wrote: Weird.. I also have joinwhenempty=no and user can still go into the queue without any agents logged in. Any ideas? Im using cvs head |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Kevin P. Fleming |Sent: Jueves, 28 de Abril de 2005 11:02 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Queues configuration | |Anton Krall wrote: | | How do you do it? I mean, if a caller is already on the queue and | suddenly all agents logoff.. How do you make the caller fall out of | the queue and into an IVR where he can leave a message? | |Have you read the sample queues.conf file? There is an option |there called 'leavewhenempty' that does exactly that. |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MEETME core uses ulaw?
Title: Normal So no one has any ideas about how to get MeetMe to work with a codec other than ulaw? Is anyone successfully doing it? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Morin Sent: Tuesday, May 03, 2005 10:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] MOH Core uses ulaw... Im trying to get Asterisk setup as a conference bridge. When I originally tried MeetMe, I was using GSM and as the conference got longer, the delay got worse and worse. From my research, I assumed that it was because MeetMe uses ulaw at its core, so everything is getting transcoded twice and each instant adds more and more delay to the cycle. To test this, I changed all of my connections to ulaw and now I get very minimal delay. However, this is not acceptable for me. Im anticipating most of my meeting attendees to come in over my VoIP connection and if this voip line is using ulaw, it will significantly reduce the number of simultaneous users that my internet connection can handle. So, it seems to me that I need to change the core codec of MeetMe to something like GSM so that I can get OK call quality, while getting the most out of my Internet connection. Does anyone know how to do this? Am I on the right track or way off with this one? Is anyone using MeetMe with GSM or any other non ulaw codec and not having a problem? Also (sorry so many questions) Im not thrilled with GSM or iLBC. I know there are a lot of people who like G.729what are the costs involved with using this one? Thanks in advance. Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemailbox on Queue?
Is there an option for a caller to quit waiting in the queue and leave a voicemail? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI timing problems: Fax Voice
On May 4, 2005 12:05 pm, Matthew Boehm wrote: May 4 10:57:04 WARNING[25650]: chan_zap.c:4394 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 2 - audio may have been lost I think that something in asterisk (not zaptel) changed in the last week to create this problem; see my last message to -dev. When the span was 0, I NEVER got that message. I haven't heard any complaints from the other office mates that use the PRI for voice, but the error just bothers me. What is the real difference between 0 and 1 on the span timing? all that the clock span means is what span * synchronizes to. clock of 0 means do not try to synchronize to the clock on this span 1 means this span is my primary clock sync source 2 means that if the span with '1' is down, use this one 3 means if the spans with 1 and 2 are down, use this one etc. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Company Signed Letter of Intent to Acquire LiveVoip, LLC
LiveVoip
has had numerous calls from some customers/brokers about this announcement. It
isa public announcement. We expect to closethis subject to all
normal conditions, in very short order. LiveVoip will remain LiveVoip LLC
operating under RV Wireless as part of a public company,with a larger
staff, more access to capital. Its our intent to setup marketingall
services in a very aggressive manner. Any questions can be addressed to: [EMAIL PROTECTED]
We are NOT raising prices, We
areNOT all leaving the company. We are going to buy more from our
vendors and continue to add more customers. There is an engineering team working
on DTMF, capacity and growth related issues. What can we say? The staff and
Management are Very Happy and look forward to working with the folks at RV
Wireless. VoiP and customer demand have made this Industry the "Next Wave" on
the Internet. Thanks to all of our great customers. In the VoIP business there
are certain to be more mergers and IPO's.
COMPANY NEWS AND PRESS RELEASES FROM OTHER SOURCES:
RV Wireless to Tap Into $200 Billion U.S. Telephone Market
as Company Signed Letter of Intent to Acquire LiveVoip, LLC., Reports IOCircuit
NOTE TO EDITORS: The Following Is an Investment Opinion Being
Issued by the IOCircuit.
LAKE HARMONY, PA, Apr 25, 2005 (MARKET WIRE via COMTEX) -- The
IOCircuit recommends RV Wireless, Inc. (OTC: RVWS), which today announced that they have signed a
letter of intent to acquire LiveVoip, LLC an Arizona-based provider of Voice
over Internet Protocol ("VoIP") communication services and products.
Jeffrey Black, President, stated, "The acquisition of LiveVoip will allow RV
Wireless to provide a complete line of VoIP products which are proprietary."
Mr. Black went on further to say: "RV Wireless is moving fast in the
development of VoIP for distribution through companies like Connectifi, Inc.
which RV Wireless owns 30% of. This acquisition will provide us with a voice
offering tailored specifically for the RV market. We can offer a voice service
that RVers will take with them lowering their cost of staying in touch."
LiveVoip also provides a platform to develop a managed service offering for the
resorts as well.
For more information, go to www.vlcn.com/rvws42505.htm
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RE: [Asterisk-Users] Asterisk CDR - Mysql
Thanks guys, it's working now. I must have missed the mysql-devel on my last build Rick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent DeShazer Sent: Wednesday, May 04, 2005 9:04 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk CDR - Mysql From your make output it looks like maybe you don't have the mySQL development package installed on this box? The one with the associated header files, etc. On the RPM-based Linux systems I've used (like Redhat, Mandrake, CentOS, Suse) the package is usually named [packagname]-devel.version.rpm, so you might search your install CD's for a mysql-devel-#.#.#.#.rpm file and try installing that (rpm -ivh [pacakgename]), then try compiling again. Brent -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rick Baranowski Sent: Wednesday, May 04, 2005 10:14 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk CDR - Mysql Matthew, Thank you very much for the help. We know that the module is not loading because we can't do the make and make install successfully for the add-ons. It's telling us that it can't find the files necessary when we do a make(print out listed below). We have renamed the add-ons dir and downloaded again from the CVS but we are still getting this error. Any thoughts? Thanks Rick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Wednesday, May 04, 2005 6:38 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk CDR - Mysql If cdr mysql status is 'command not found' then that means you haven't loaded the module. Check your module path to make sure it really is there. (/usr/lib/asterisk/modules/) If it is indeed there, do load cdr_addon_mysql.so from CLI* You might want to check modules.conf and make sure you have an autoload in there. -Matthew Rick Baranowski wrote: We seem to be having the same problem. The cdr command is not found, so we tried to do a make and install on the add-ons but it can't see to find the files when we run 'make clean make make install'. We have downloaded from CVS and the files look to be there but it still can't find the files. Could someone help? Thanks Rick [EMAIL PROTECTED] asterisk-addons]# make clean make make install rm -f *.so *.o .depend make -C format_mp3 clean make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' rm -f *.o *.so *~ make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` app_addon_sql_mysql.c:31:19: mysql.h: No such file or directory cdr_addon_mysql.c:33:19: mysql.h: No such file or directory cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o common.o common.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o dct64_i386.o dct64_i386.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o decode_ntom.o decode_ntom.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o layer3.o layer3.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o tabinit.o tabinit.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o interface.o interface.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o format_mp3.o format_mp3.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -shared -Xlinker -x -o format_mp3.so common.o dct64_i386.o decode_ntom.o layer3.o tabinit.o interface.o format_mp3.o make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3' for x in format_mp3/format_mp3.so ; do install -m 755 $x /usr/lib/asterisk/modules ; done [EMAIL PROTECTED] asterisk-addons]# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Monday, May 02, 2005 7:03 PM To: Asterisk Users Subject: Re: [Asterisk-Users] Asterisk CDR - Mysql If you enabled it in logger.conf, it should be at /var/log/asterisk/debug What does cdr mysql status do? If it says no such command then you haven't loaded the cdr module. Did you do make install inside the
RE: [Asterisk-Users] bri error
Hi David I was on site with this system and saw some other error something like this: Avoided deadlock on zap 1-1 chan_lock. maximum retries 10 This came up between the errors: May 3 17:43:12 pbxct kernel: qozap: dropped audio card 1 cardid 0 bytes 19 z1 71 z2 36 If you call into the system on chan 1-1 call goes dead and when calling out on chan 1-1 it will work intermittently but tends to drop the call during conversation. We have had the telco out to test the lines and they are sure there is no problem on there side. [hhmmm] Seems to me like a telco problem, what do you think? regards doug -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of David Masure Sent: Friday, April 29, 2005 12:20 PM To: [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] bri error The problem may then originate from the NT of your telco -Message d'origine- De : Altus Snyman [mailto:[EMAIL PROTECTED] Envoy : vendredi 29 avril 2005 12:21 : David Masure Cc : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: [Asterisk-Users] bri error if I do a zttool it shows TE mode On Fri, 2005-04-29 at 12:14, David Masure wrote: Did you put your card in TE mode ? To it seems you have configured your card to act like a NT but if you are connected to bri telco lines, it should be in TE mode check in your zaptel.conf : bri te signalling regards David -Message d'origine- De : Altus Snyman [mailto:[EMAIL PROTECTED] Envoy : vendredi 29 avril 2005 12:08 : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] bri error Good day all This is a error that keeps on popping up in my /var/log/messages when I get incoming or outgoing calls on my bri card connected to 4 telco isdn units?It is a junghanns 4 port card with the latest version of the drivers and latest asterisk Apr 29 11:37:39 ccv kernel: qozap: BAD CRC for hdlc frame on card 1 (cardID 0) S/T port 1 Apr 29 11:37:39 ccv kernel: qozap: check the 100 Ohm termination for this span! Apr 29 11:45:25 ccv kernel: zaptel Disabled echo canceller because of tone (rx) on channel 1 Please help and advice? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Attended Transfer using wrong Context
The phone's context is cytel-internal. This allows us to hit 3XXX to get someone on the inside. If you hit 9 at the beginning, you Goto() the cytel-outgoing context. So lets make a call..I'll dial 918005551212 (toll free directory). The 9 sends it to cytel-outgoing. Call is made. Bridged. I then hit #9 for attended transfer. Allison says Transfer. I start to enter 3013. But right after I hit the first 3, it returns failed transfer: res_features.c:800 builtin_atxfer: Did not read data. Wtf? So I do it again; and again. I tried every number and they all returned the same error. But this time I press 93013 and the call goes out the cytel-outgoing context. ???!?? I'm lost. What is this thing doing? Being very bad. Just some ideas: What's the transferdigittimeout setting in features.conf? Maybe it's not giving you enough time to really enter an extension. Also, what happens when you change attended transfer to something other than #9 - Noah ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960: Builtin CFwdAll working?
I turned off reinvite and I still get the same behavior. What is your promiscredir set at? -Matthew Alexander Lopez wrote: It works for me. Do you have reinvites enabled. I do not. That may explain why * is sending a redirect. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Wednesday, May 04, 2005 11:35 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960: Builtin CFwdAll working? Hey guys, Lots of nice people on the list using 7960s and using/discovering features that I didn't think possible. (Re: Multi Line Appearance). Wanted to know if anyone has gotten the 'CFwdAll' button to properly work. The problem I am seeing is that if someone presses the button and types in their cell (for instance), I get a local channel into a loop message in asterisk. Here is an incomming call to my DID to my 7960 (x3044) -- Called 3044 -- Got SIP response 302 Moved Temporarily back from 10.0.0.36 -- Now forwarding Zap/1-1 to 'SIP/[EMAIL PROTECTED]:5060' (thanks to SIP/3044-a649) -- Got SIP response 482 Loop Detected back from 10.0.3.10 -- Now forwarding Zap/1-1 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/10.0.3.10:5060-a32b) May 4 10:28:21 NOTICE[25650]: chan_local.c:436 local_alloc: No such extension/context [EMAIL PROTECTED] creating local channel May 4 10:28:21 NOTICE[25650]: app_dial.c:355 wait_for_answer: Unable to create local channel for call forward to 'Local/[EMAIL PROTECTED]' (cause = 0) The problem is, that the phone makes a brand new SIP call to asterisk. Well, all incomming calls go into the all-incomming context. I was expecting the new call to use the same context that the phone is registered into. Any thoughts? Ideas? Thanks, Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Aastra 480i
Hi, I have an Aastra 480i running with Asterisk. I can make local and long distance calls on it no problem, but if I dial a number where another phone system is involved and I need to punch in some numbers, this is no go! I can hit all the numbers on the phone that I want and nothing happens. This doesn't happen with a Snom phone or X-lite so this rules out problems with everything EXCEPT the 480i. Anyone else have this problem or know a possible solution? Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newer Dell Servers + TDM card
Folks, This is a firmware bug in the TDMxxx and TExxx cards that Digium has recently fixed. I did an advanced replacement for mine which involved me buying another one and them refunding me when they got my old one back. Get onto their tech support. Mark Matt Schulte wrote: Is this with the TDM400P card right? -Original Message- From: David Brodbeck [mailto:[EMAIL PROTECTED] Sent: Monday, May 02, 2005 2:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Newer Dell Servers + TDM card -Original Message- From: Matt Schulte [mailto:[EMAIL PROTECTED] Really, how long does it take to recover? Mine just totally locks. No time at all. The only reason I know an NMI occurs is the front panel light, and the Dazed and confused, but trying to continue message from the kernel. I'm using a Dell PowerEdge 800. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
