Re: [asterisk-users] missing chan_zap.so
On Wed, Apr 11, 2007 at 08:09:16PM -0700, Yuan LIU wrote: From: Sanjay Rajdev [EMAIL PROTECTED] Date: Thu, 12 Apr 2007 01:29:51 +0530 (IST) [good stuff sniffed] and downloaded zaptel 1.4.1, after that executed the following commands ./configure make clean make make install Went to asterisk folder ./configure make clean make make upgrade But could not get chan_zap.so then did the make install of asterisk. still missing the chan_zap.so Have you loaded wctdm? Whatever kernel modules are loaded does not matter to the build of chan_zap.so Do you have: Should be generated by 'make': channels/chan_zap.so # under the asterisk build directory Should be copied by 'make install': /usr/lib/modules/chan_zap.so -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to set fromuser in sip.conf so each user gets it's own callerid?
I'm a first time user of Asterisk and have a working setup which I find
clumsy. How can I clean things up to make the dialplan easier to maintain?
My problem
==
I have 6 public numbers that can reach 6 individual users. I have 6
lines like this in sip.conf:
[general]
register = 31307115622:[EMAIL PROTECTED]/622
register = 31307115627:[EMAIL PROTECTED]/627
each user registers with something like this:
[siemens1](xanadu-internal)
type=friend
callerid=Theo Band
context=xanadu-thba
[belcentrale-out-thba](belcentrale-outgoing)
type=peer
fromuser=31307115622
My extenson.conf looks like this:
[xanadu-thba]
exten = _+.,1,goto(00${EXTEN:1},1);00 is long distance calls = +
exten = _0[1-9].,1,goto(0031${EXTEN:1},1);local calls =0031
exten = _0031Z., 1,Macro(dialout,SIP/[EMAIL PROTECTED]);NL
exten = _+ZXX., 1,Macro(dialout,SIP/00${EXTEN:[EMAIL PROTECTED]);INT
My dialplan contains a context like xanadu-user for every user, only
to be able to set the fromuser correctly.
This works but I prefer to have one dialplan and set some sort of
variable containing the fromuser in sip.conf for every registered user.
I read the entire march list (that's a lot!) and was also not able to
find a proper search term that covers this question.
Thanks,
Theo
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[asterisk-users] hanguponpolarityswitch - where did it go??
There are a few mentions in the wiki [1] about a zapata.conf flag hanguponpolarityswitch. It is meant to cause Asterisk to detect a hangup when the line polarity switches at the end of the call. The wiki mentions using the flag in zapata.conf but when I do Asterisk ignores it: Apr 12 17:59:38 WARNING[12804]: chan_zap.c:10875 setup_zap: Ignoring hanguponpolarityswitch Does anyone have any ideas how to enable or use this feature? Many thanks, Nick. [1] http://www.voip-info.org/wiki/view/Australia+Asterisk+Details ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hanguponpolarityswitch - where did it go??
Nick Adams wrote: There are a few mentions in the wiki [1] about a zapata.conf flag hanguponpolarityswitch. It is meant to cause Asterisk to detect a hangup when the line polarity switches at the end of the call. The wiki mentions using the flag in zapata.conf but when I do Asterisk ignores it: Apr 12 17:59:38 WARNING[12804]: chan_zap.c:10875 setup_zap: Ignoring hanguponpolarityswitch Does anyone have any ideas how to enable or use this feature? Hi, as far as I know, it only says ignoring when you do a reload, as Asterisk is telling you its not reconfiguring this variable, to change it you might need a restart. So hanguponpolarityswitch only gets looked at on startup, not reloads. -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] compile problem with wavelenght
Hello Im trying to install an old version of Asterisk. But it isnt working: when I run make install: gcc -o gentone gentone.c -lm ./gentone busy 480 620 Wavelength 1 (in samples): 16.7 Minimum samples (1): 50 (3.00.3 wavelengths) Wavelength 1 (in samples): 12.90323 Minimum samples (1): 400 (31.00.3 wavelengths) Need 400 samples Wrote busy.h ./gentone ringtone 440 480 Wavelength 1 (in samples): 18.18182 Minimum samples (1): 200 (11.00.3 wavelengths) Wavelength 1 (in samples): 16.7 Minimum samples (1): 50 (3.00.3 wavelengths) Need 200 samples Wrote ringtone.h gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -fomit-frame-pointer -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -fPIC-c -o chan_oss.o chan_oss.c gcc -shared -Xlinker -x -o chan_oss.so chan_oss.o -ldl -lpthread -lncurses -lm -lresolv -lssl gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -fomit-frame-pointer -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -fPIC-c -o chan_phone.o chan_phone.c chan_phone.c:41:29: error: linux/compiler.h: No such file or directory make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/services/asterisk/asterisk-1.2.10/channels' make: *** [subdirs] Error 1 [EMAIL PROTECTED] asterisk-1.2.10]# Whats happening? I already tried with 3 different versions downloaded from asterisk.org site. Thanks Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test
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Re: [asterisk-users] test
ACK 2007/4/12, Razza [EMAIL PROTECTED]: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alberto Sagredo RD area Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Office phone : +34 91 120 5080 Direct phone : +34 91 120 50 39 Peoplecall Network : 700 757 139 Fax number : +34 91 661 9460 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] compile problem with wavelenght
On Thu, Apr 12, 2007 at 10:25:37AM +0100, Joao Pereira wrote: Hello Im trying to install an old version of Asterisk. But it isnt working: when I run make install: gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -fomit-frame-pointer -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -fPIC-c -o chan_phone.o chan_phone.c chan_phone.c:41:29: error: linux/compiler.h: No such file or directory make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/services/asterisk/asterisk-1.2.10/channels' make: *** [subdirs] Error 1 This is a known problem that has been fixed in later versions of asterisk 1.2 . Alternatively, build the same version withough building chan_phone.so . -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Hello , iam having 6 asterik cards on three different servers I am using asterisk 1.2.14 with zaptel 1.2.12 on fedora core 5 (2.6.17.1). now every 3 days i need to rmmod/modprobe wctdm driver to detect the call. callers wont get the IVR prompt. after rmmod and modprobe the wctdm ..it works fine. earlier i wanted to restart every day...so i removed the CLI detection on British telecom line.now its happening evry 3 days..i have used this part in additon to normal config. but it gave that error..everyday..(asterisk didnt detect the incming call) then i remove this part...now that happeing evry 3 days..(ima connected to British telecom PSTN). i have enabled loadzone=uk.. usecallerid=yes cidsignalling=v23 cidstart=polarity this is my zaptel config.. (NO CLI detection enabled) signalling=fxs_ks busydetect=yes busycount=8 threewaycalling=yes group=1 context=sip echocancel=yes channel= 1-8 echocancelwhenbridged=yes echotraining=20 echotraining=yes dtmfmode=rfc2833 rxgain=4.0 txgain=4.0 My fxo cards are connected to British telecom . can it be a problem with BT singnaling..?? because asterisk verisn 1.07 worked without any erros.. or can it be a problem with the card ? many thanks, Tharanga ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Automatic Hang
Hi guys!
Im using Asterisk 1.2 with mISDN support.
I have problems with Pickup calls with my Grandstream Buttons . I set up on
Dial Plan this:
Exten = _**XXX,1,Pickup(SIP/{EXTEN:2}) but it doesnt work if the call
comes from mISDN. So, I wanna do something to this:
Exten = _**XXX,1,SendDtmf(*8#) because if I introduce *8# into my telephone
i can pickup a call from everywhere. BUT the problem is that I cannot dial
automatically *8#. Does anybody know how to do it?
THANKS
Saludos, Lukassky.
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[asterisk-users] Asterisk 1.2.14 and zaptel 1.2.12 ivr hangs every 2 days
Hello , iam having 6 asterik cards on three different servers I am using asterisk 1.2.14 with zaptel 1.2.12 on fedora core 5 (2.6.17.1). now every 3 days i need to rmmod/modprobe wctdm driver to detect the call. callers wont get the IVR prompt. after rmmod and modprobe the wctdm ..it works fine. earlier i wanted to restart every day...so i removed the CLI detection on British telecom line.now its happening evry 3 days..i have used this part in additon to normal config. but it gave that error..everyday..(asterisk didnt detect the incming call) then i remove this part...now that happeing evry 3 days..(ima connected to British telecom PSTN). i have enabled loadzone=uk.. usecallerid=yes cidsignalling=v23 cidstart=polarity this is my zaptel config.. (NO CLI detection enabled) signalling=fxs_ks busydetect=yes busycount=8 threewaycalling=yes group=1 context=sip echocancel=yes channel= 1-8 echocancelwhenbridged=yes echotraining=20 echotraining=yes dtmfmode=rfc2833 rxgain=4.0 txgain=4.0 My fxo cards are connected to British telecom . can it be a problem with BT singnaling..?? because asterisk verisn 1.07 worked without any erros.. or can it be a problem with the card ? many thanks, Tharanga ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Measuring audio file legth
Hi,
I have to set call length to 3min, but before hangup have to warn
caller. There are many IVRmenu and submenu options with different
warning audio.
I have to measure somehow the audio file length and subtract it from 3
minutes.
exten = _X.,1,Set(_AudioLegth=MeasureAudioLength(WarningAudioX))
exten = _X.,n,Set(TIMEOUT(absolute)=${${MaxCallDuration}-${AudioLength}})
Any idea?
Thank you!
--
Suich
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Re: [asterisk-users] compile problem with wavelenght
Hello Thanks a lot for the help. I just commented these lines and its working: #ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/include/linux/ixjuser.h)$(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/ixjuser.h),) # CHANNEL_LIBS+=chan_phone.so #endif I just hope that this doesnt bring me problems in the future :P Thanks regards Joao Pereira Tzafrir Cohen wrote: On Thu, Apr 12, 2007 at 10:25:37AM +0100, Joao Pereira wrote: Hello Im trying to install an old version of Asterisk. But it isnt working: when I run make install: gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -fomit-frame-pointer -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -fPIC-c -o chan_phone.o chan_phone.c chan_phone.c:41:29: error: linux/compiler.h: No such file or directory make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/services/asterisk/asterisk-1.2.10/channels' make: *** [subdirs] Error 1 This is a known problem that has been fixed in later versions of asterisk 1.2 . Alternatively, build the same version withough building chan_phone.so . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum retries exceeded on transmission
I confirm the same behaviour. I use asterisk with Mera Softswitch (with SIP HIT). After upgrading from 1.2.13 to 1.2.14 Maximum retries exceeded... messages began to appear in logs. About 10% of calls were lost. I've dumped such calls and don't see anything suspicous in Mera's packets. Asterisk doesn't reply for first several INVITEs from Mera but then it replies OK, Mera sends back ACK but it seems that asterisk ignores it and tries to send OK. After trying to send OK several times asterisk hangs up the call. I've attached the text file where this can be seen. Mera SS is 10.150.16.4. We see that asterisk replies to INVITEs after 4 seconds. That's wierd. Server is not heavily loaded - about 10 simultanious calls. I've downgraded to 1.2.13 and problem has gone away. I guess there is something wrong with asterisk. Regards. Andrey Solovjov. Edoardo Serra: Same to me !! Calls from OpenSER to Asterisk It happens only with Asterisk versions = 1.2.14 I'm going to capture some traffic Tnx for help Regards Alex Balashov ha scritto: Joao, It sounds like the proxy is not acknowledging the Asterisk's processing of the INVITE, but I could be wrong. It would be helpful to supply a packet capture between OpenSER and Asterisk so we could see the setup flow. Thanks, -- Alex On Tue, 10 Apr 2007, Joao Pereira said something to this effect: Hello My asterisk is receiving calls from OpenSER but all calls hangup in 20 seconds. This only happens because Im using Asterisk2Billing's AGI (without A2Billing it doesnt hang up). does someone knows whats the problem?? Here is my Asterisk debug: (xxx.xxx.xxx.xxx - the phone's IP) Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: Unable to spawn mp3player Apr 10 02:04:18 NOTICE[8349]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 12282 (Critical Response) Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1245 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. Apr 10 02:06:01 NOTICE[8360]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Thanks for the help Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users |Time | 10.150.16.4 | 10.153.144.131| |488,548 | INVITE SDP ( telephone-event) |SIP From: sip:[EMAIL PROTECTED] To:sip:[EMAIL PROTECTED]:5060 | |(5060) -- (5060) | |489,047 | INVITE SDP ( telephone-event) |SIP From: sip:[EMAIL PROTECTED] To:sip:[EMAIL PROTECTED]:5060 | |(5060) -- (5060) | |489,539 | INVITE SDP ( telephone-event) |SIP From: sip:[EMAIL PROTECTED] To:sip:[EMAIL PROTECTED]:5060 | |(5060) -- (5060) | |490,544 | INVITE SDP ( telephone-event) |SIP From: sip:[EMAIL PROTECTED] To:sip:[EMAIL PROTECTED]:5060 | |(5060) -- (5060) | |492,429 | 100 Trying| |SIP Status | |(5060) -- (5060) | |492,434 | 200 OK SDP ( telephone-event) |SIP Status | |(5060) -- (5060) | |492,435 | RTP (g711A) |RTP Num packets:845 Duration:19.980s ssrc:858984592 | |(21816) -- (16296) | |492,448 | ACK | |SIP Request | |(5060) -- (5060) | |492,557 | 200 OK SDP ( telephone-event) |SIP Status | |(5060) -- (5060) | |492,560 | ACK | |SIP Request | |(5060) -- (5060) | |492,683 | 200 OK SDP ( telephone-event) |SIP Status | |(5060) -- (5060) | |492,708 | ACK | |SIP Request | |(5060) -- (5060) | |492,834 | RTP (g711A) |RTP Num packets:11 Duration:0.206s ssrc:1605848118 | |(21816) -- (16296) | |493,050 | 200 OK SDP ( telephone-event) |SIP Status | |(5060) -- (5060) | |493,051 | RTP
[asterisk-users] Re: Which SIP phones to buy?
Quoting Stephen Bosch [EMAIL PROTECTED]: I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I don't want to broaden my knowledge. Advice, anyone? -Stephen- You said 'office' so I'm presuming you want business quality. If you have already tried the Polycom's I'd look at Aastra (just did a 50+ seat implementation with 9133i's 480i's) and also look at the Cisco 79xx's. Cisco's Aastra's both handle multiple appearances differently but both are excellent. Cisco has superb handsfree quality. Aastra has better BLF support. You will have to evaluate for yourself. Aastra is significantly cheaper. That said, there is a 7960 on my desk that isn't going anywhere soon. I hear the Grandstream firmware is better now but physically they are still pretty flimsy. I would stay away from them for anything but experimentation. dbc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR(disposition)
Hello to everybody, I have a problem with the disposition filed that asterisk write in mysql table. What I notice is that for every outbound calls (for example to a mobile phone) I see in disposition field the string ANSWERED when I reject the call and also when I really answer the call, while in the variable DIALSTAUS I have the correct status of the call (BUSY, CHANUNAVAIL, ANSWERED, NO ANSWER etc). Can anyone help me? Bye, Damiano Bertuna. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verizon-Vonage Lawsuit
You hit the nail on the head. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Wednesday, April 11, 2007 2:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verizon-Vonage Lawsuit Salvatore Giudice wrote: BTW, the main problem with these patents is that they tend to lower the rate of adoption for new standards. Nothing kills a standard quicker than when someone patents it. For example, someone out there even has a patent on ENUM: http://www.freepatentsonline.com/20060020713.html?highlight=enumstemming=on It made me mad that he beat me to it. Roflol... Regardless, this won't help with ENUM adoption. Any joker with about $6k per patent and some time on his hands to monitor emerging standards can easily generate some patent entertainment for themselves at the expense of others... So, the question of the day is: Have you thought about patenting something today? It's easy. I just got a new idea while writing this for an ENUM related patent that I may pursue at some point... =) The US patent system is totally broken. It started with lobbying efforts to relax the applicability rules for patents for short-term gain. In the long term, it's going to do big damage to American competitiveness. And that's the sad thing about this. It discourages actual innovation (despite Wall Street protests to the contrary). If everytime you want to build on somebody else's work you have to build a skein of licencing agreements, you start to ask yourself, why should I bother? More and more companies are answering that one with We shouldn't -- there's enough action to be had in other parts of the world, where the conditions are much less onerous. Another example of that kind of short-sighted thinking is what happened to the US crypto business when all the export controls were brought in. (A lot of damage was done in exchange for no demonstrable security benefit.) Obviously, a market that big and moneyed isn't going to be ignored: how can it be? But what used to be a no-brainer isn't so obvious anymore -- staying out of the US market is a serious option where it wasn't before, and that just leads to further Balkanization. It's fitting that an open source product like Asterisk is helping keep the US in the game. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR(disposition)
I think this has to do with how your dial plan is setup. If you are making a call to a cell phone, i'm assuming that you are using an FXO (or some sort of phone service). My guess is that the disposition is being marked ANSWERED because the FXO is picking up (or the phone service is) and answering the call from Asterisk. The Dial() function is communicating with the FXO to determine whether or not the call is actually working. Unfortunately, my guess is that the CDR is only applicable to the connection between Asterisk and your termination point (FXO or otherwise). I say this because I know that, in the instance of QOS statistics, the CDR would not be able to know whats happening beyond the FXO. On 4/12/07, damiano bertuna [EMAIL PROTECTED] wrote: Hello to everybody, I have a problem with the disposition filed that asterisk write in mysql table. What I notice is that for every outbound calls (for example to a mobile phone) I see in disposition field the string ANSWERED when I reject the call and also when I really answer the call, while in the variable DIALSTAUS I have the correct status of the call (BUSY, CHANUNAVAIL, ANSWERED, NO ANSWER etc). Can anyone help me? Bye, Damiano Bertuna. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verizon-Vonage Lawsuit
I blogged about it here http://deancollinsblog.blogspot.com/2007/04/software-patents.html Though I think GigaOm nailed it when they wrote Verizon can't make the Internet go away with a patent lawsuit. http://gigaom.com/2007/04/08/voip-patent-mess/ Cheers, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Wednesday, April 11, 2007 2:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verizon-Vonage Lawsuit Salvatore Giudice wrote: BTW, the main problem with these patents is that they tend to lower the rate of adoption for new standards. Nothing kills a standard quicker than when someone patents it. For example, someone out there even has a patent on ENUM: http://www.freepatentsonline.com/20060020713.html?highlight=enumstemmin g=o n It made me mad that he beat me to it. Roflol... Regardless, this won't help with ENUM adoption. Any joker with about $6k per patent and some time on his hands to monitor emerging standards can easily generate some patent entertainment for themselves at the expense of others... So, the question of the day is: Have you thought about patenting something today? It's easy. I just got a new idea while writing this for an ENUM related patent that I may pursue at some point... =) The US patent system is totally broken. It started with lobbying efforts to relax the applicability rules for patents for short-term gain. In the long term, it's going to do big damage to American competitiveness. And that's the sad thing about this. It discourages actual innovation (despite Wall Street protests to the contrary). If everytime you want to build on somebody else's work you have to build a skein of licencing agreements, you start to ask yourself, why should I bother? More and more companies are answering that one with We shouldn't -- there's enough action to be had in other parts of the world, where the conditions are much less onerous. Another example of that kind of short-sighted thinking is what happened to the US crypto business when all the export controls were brought in. (A lot of damage was done in exchange for no demonstrable security benefit.) Obviously, a market that big and moneyed isn't going to be ignored: how can it be? But what used to be a no-brainer isn't so obvious anymore -- staying out of the US market is a serious option where it wasn't before, and that just leads to further Balkanization. It's fitting that an open source product like Asterisk is helping keep the US in the game. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 301 questions
Stephen Bosch wrote: Lee Jenkins wrote: Hi all, I just purchased a Polycom 301 for my home office and I believe I have it setup correctly as I can dial out, receive calls in, etc. However, I'm having the following issue: When calling a local number over a Zap line, I hear a lot of feed back on the line. I had a Grandstream configured with the same information before I got the 301 and never had that kind of feedback noise. Feedback? As in high-pitched squealing? I can't imagine any circumstance under which a SIP phone would even allow feedback. Maybe you mean echo? You might also be talking about sidetone, which is the portion of your own voice that the phone pipes back to you so that you can adjust your voice accordingly. Sidetone can be set in the phone configuration; before you do that, though, I need to know what you mean by feedback. Sorry, should have been more detailed. It's a sort of background humming noise, almost like that if you placed the phone next to a high output electric device. Didn't hear it with Grandstream phone so I thought it may be something I can change in Polycom. Of course, it could also be that polycom is better phone, but more apt to pick up background noise that It's only on zap calls. I'm still playing with rxgain and txgain in hopes of resolving this. Basically, the humming noise increase with volume. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] HPEC audio clipping
No luck yet. No response from Digium support so I guess that they are still waiting for the Zaptel test code. Greg -Original Message- From: Stephen Bosch [mailto:[EMAIL PROTECTED] Sent: Thursday, 12 April 2007 4:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HPEC audio clipping Greg Siemon wrote: Thanks for the helps Stephen. I was running non standard gains but setting regain and txgain to zero (then reloading chan_zap.so) does not help. I still get the broken audio, in fact sometimes I don't get any audio at all. In testing the server just froze a number of times and had to be rebooted via the power switch. I am using the latest Zaptel 1.2.16 files and the latest fxotune from the 1.4 release and I still see this issue. Very interested to get this working but without the HPEC my server is rock solid (only have to reboot it when I install kernel updates). I don't believe it is my system but am happy to do any testing others may suggest. Have you had any luck with this, Greg? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
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Re: [asterisk-users] Maximum retries exceeded on transmission
Hello Thanks a lot for your reply. Im now using asterisk-1.2.10 and the problem disappeared. Thanks regards Joao Pereira Edoardo Serra wrote: Same to me !! Calls from OpenSER to Asterisk It happens only with Asterisk versions = 1.2.14 I'm going to capture some traffic Tnx for help Regards Alex Balashov ha scritto: Joao, It sounds like the proxy is not acknowledging the Asterisk's processing of the INVITE, but I could be wrong. It would be helpful to supply a packet capture between OpenSER and Asterisk so we could see the setup flow. Thanks, -- Alex On Tue, 10 Apr 2007, Joao Pereira said something to this effect: Hello My asterisk is receiving calls from OpenSER but all calls hangup in 20 seconds. This only happens because Im using Asterisk2Billing's AGI (without A2Billing it doesnt hang up). does someone knows whats the problem?? Here is my Asterisk debug: (xxx.xxx.xxx.xxx - the phone's IP) Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: Unable to spawn mp3player Apr 10 02:04:18 NOTICE[8349]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 12282 (Critical Response) Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1245 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. Apr 10 02:06:01 NOTICE[8360]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Thanks for the help Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
You seem to have misplaced your message/comment/question. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Play audio and continue to next priority before audio ends...
In article [EMAIL PROTECTED], Gordon Henderson [EMAIL PROTECTED] wrote: On Wed, 11 Apr 2007, Tony Mountifield wrote: Alejandro Mejía [EMAIL PROTECTED] wrote: I would like to know how to playback an audio file to the caller, and while it's played asterisk to continue executing the next priorities on extensions.conf That's not the case when using playback command, because the next priority is executed until the audio file ends playing. I want to evaluate some variables while caller hears the audio file. Any ideas? Look at the Background() application. It does just what you are asking for. I'm surprised no-one else has mentioned this. Are you sure it does that? Hmmm, I thought I was, but it looks like I was mistaken... I was probably misled by the name Background(), which is perhaps not an accurate description of its function then. I'm under the impression that it waits until the sound(s) have finished playing before moving on to the next priority. (While listening for digits to be pushed, then be dialled) So the only difference between Playback() and Background() is that the latter will accept incoming digits (and use them to divert the dialplan) and the former won't. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING
moises, guys, just an update, steve released new packages early april. i just did a successful compile, tomorrow i will test with a live e1 line. i managed to compile it with asterisk-1.4.2 a series of patches is on the way after a successful test. [EMAIL PROTECTED] wrote: nivlekch, nice to hear that :) I hope more people can test this. On 3/14/07, nivlekch [EMAIL PROTECTED] wrote: nice job moises, the hardwork you and steve put into chan_unicall is remarkable. with a little editing and tweaking, i was able to make the port to 1.4 here in the philippines without any problems. some part of libmfcr2 has to be changed for proper/better ANI exchage with PLDT(telco). looking good so far, better than the experience in 1.2, i'll post any update soon. anybody interfacing with PLDT interested, email me offline. [EMAIL PROTECTED] wrote: Im glad to let you know that finally I invested some time to make work Unicall in Asterisk 1.4, I must say not much testing could be done since I have no hardware available ( cards, servers ), however a friend was able to test it with a couple of calls with success, I need you to test this and report some feedback. The sources are available in: http://moy.ivsol.net/unicall/soft-switch/r1b1/ Kind Regards Moises Silva ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to install asterisk on redhat ?
On Wed, Apr 11, 2007 at 07:32:52PM +0300, Dovid B wrote: I wrote this ages ago. You may want to get more current software than the URL's that are listed. #YUM INSTALLS yum -y install gcc yum -y install kernel-source actually: kernel-devel (or kernel-smp-devel) yum -y install bison yum -y install doxygen yum -y install openssl-devel yum -y install flex yum -y install gcc # WGET DOWNLOADS FROM H6315 / TARBALLS wget http://www.h6315.com/pub/asterisk/asterisk-1.0.6.tar.gz wget http://www.h6315.com/pub/zaptel/zaptel-1.0.6.tar.gz wget http://www.h6315.com/pub/libpri/libpri-1.0.6.tar.gz wget http://www.h6315.com/pub/openssl/openssl-0.9.7f.tar.gz wget http://www.h6315.com/pub/apache/httpd-2.0.53.tar.gz wget http://www.h6315.com/pub/mysql/mysql-standard-4.1.10a-pc-linux-gnu-i686.tar.gz wget http://www.h6315.com/pub/php/php-4.3.10.tar.gz wget http://www.h6315.com/pub/mpg123/mpg123-0.59r.tar.gz What is the point in rebuilding stuff that is already availble from your distribution? And is actively mintained by it? I hope whoever installed by such a tutorial is not still using those obsolete versions. # UNTAR EVERYTHING tar -xvzf asterisk*.tar.gz tar -xvzf zaptel*.tar.gz tar -xvzf libpri*.tar.gz tar -xvzf openssl*.tar.gz tar -xvzf httpd*.tar.gz tar -xvzf mysql-*.tar.gz tar -xvzf php*.tar.gz tar -xvzf mpg123*.tar.gz rm -f *.tar.gz rm -f *.rpm # INSTALL OPEN SSL cd /usr/src/openssl* ./config make make test make install # INSTALL APACHE cd /usr/src/httpd-2* ./configure --prefix=/wwwroot --enable-so --enable-rewrite --enable-headers --enable-expires -enable-deflate --with-z --enable-speling --enable-ssl make make install # INSTALL MYSQL cd /usr/src mv mysql* /usr/local cd /usr/local groupadd mysql useradd -g mysql mysql ln -s mysql-standard-4.1.10a-pc-linux-gnu-i686 mysql cd mysql scripts/mysql_install_db --user=mysql chown -R root . chown -R mysql data chgrp -R mysql . cp support-files/mysql.server /etc/init.d chmod +x /etc/init.d/mysql.server ln -s /usr/local/mysql/bin/mysql /usr/bin/mysql # INSTALL PHP cd /usr/src/php* ./configure --prefix=/wwwroot/php --with-apxs2=/wwwroot/bin/apxs --with-config-file-path=/wwwroot/php --with-mysql --enable-gd --with-mysqli=/usr/local/mysql/bin/mysql_config make make install # INSTALL MPG123 cd /usr/src/mpg123* make linux make install # INSTALL ZAPTEL cd /usr/src/zap* perl -pi~ -e 's/# ztdummy/ztdummy/' Makefile make clean make make install # INSTALL LIBPRI cd /usr/src/libp* make make install #INSTALL ASTERISK cd /usr/src/aster* make clean make make install make samples make progdocs Have some mercy on the CPU and HD, and spare this one... Tzafrir, I wrote this a long time ago (as you can see from the asterisk version that I was using). It was specific for a VPS server that I was using. I posed that the URL's were not up to date. It was more of a guide for him. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 501 issue with latest firmware: sluggish keys
Exactly. It's a weird issue, and I can't imagine what the problem is, except maybe for bad phones (but then again, why would the phones be only bad with 2.x?) UnlessI have bootrom 3.2.2.0019. Is that what people running thelatest have? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Thursday, April 12, 2007 00:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 501 issue with latest firmware: sluggish keys It has nothing to do with actually dialing. Even trying to press end call or the speakerphone button does not work at times. Have tried removing side cars etc, but definately seems to be a bug in the 2.x code stream. On Apr 11, 2007, at 5:37 PM, Eric ManxPower Wieling wrote: Jim King wrote: I've seen an issue like this from time to time on 601s, even with the latest firmware. Not just the softkeys, but also the dial keys. The phones seem to run slow sometimes, failing to respond to a key press right away but getting to it eventually. It usually clears up after a few seconds. Also, I've noticed that the 601s sometimes ignore key presses altogether, just as you describe. I have not yet found a solution for this problem... Try setting this in sip.cfg: dialplan.impossibleMatchHandling=1 I suspect it is either 0 or 2 now. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which SIP phones to buy?
Stephen Bosch wrote: Stephen Bosch wrote: I need to buy some new phones for our own offices. I've used only Polycom phones until now, but I'd like to broaden my experience. I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I don't want to broaden my knowledge... ...because I like to stay dumb. Of course, that's not what I meant :) We have Cisco, Aastra 480i and Grandstream GXP2000 phones in house. I only recommend the Cisco phones to people I don't like, overpriced and far too much work. The Aastra 480i is a good quality phone, on par with Cisco and probably with Polycom (though I've never used them). Voice quality is good, phone feels robust. Config is well documented and contained in two text files (one global, one MAC specific). Good web interface on the phone. Aastra support have been very responsive. Grandstream phones are lower quality but good value for money. Sound and feel of phones is not so good as Aastra or Cisco. Configuration is through a binary file, a bit fiddly, but quite manageable with a few scripts. Good web interface on the phone. Grandstream support have also been very responsive. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING
Hi Nivlekch, Thanks for that, just a comment: What do you mean by new packages? new for spandsp, libmfcr2, unicall? chan_unicall? On 4/12/07, nivlekch [EMAIL PROTECTED] wrote: moises, guys, just an update, steve released new packages early april. i just did a successful compile, tomorrow i will test with a live e1 line. i managed to compile it with asterisk-1.4.2 a series of patches is on the way after a successful test. [EMAIL PROTECTED] wrote: nivlekch, nice to hear that :) I hope more people can test this. On 3/14/07, nivlekch [EMAIL PROTECTED] wrote: nice job moises, the hardwork you and steve put into chan_unicall is remarkable. with a little editing and tweaking, i was able to make the port to 1.4 here in the philippines without any problems. some part of libmfcr2 has to be changed for proper/better ANI exchage with PLDT(telco). looking good so far, better than the experience in 1.2, i'll post any update soon. anybody interfacing with PLDT interested, email me offline. [EMAIL PROTECTED] wrote: Im glad to let you know that finally I invested some time to make work Unicall in Asterisk 1.4, I must say not much testing could be done since I have no hardware available ( cards, servers ), however a friend was able to test it with a couple of calls with success, I need you to test this and report some feedback. The sources are available in: http://moy.ivsol.net/unicall/soft-switch/r1b1/ Kind Regards Moises Silva ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 501 issue with latest firmware: sluggish keys
Somehow, I ended up with BootROM 3.2.3.0002 (which as far as I can tell hasn't been released yet...) and SIP version 2.1.0.2708. I do see the sluggish buttons from time to time. Rarely, but I do see it. --TS Mike [EMAIL PROTECTED] 4/12/2007 9:59 AM Exactly. It's a weird issue, and I can't imagine what the problem is, except maybe for bad phones (but then again, why would the phones be only bad with 2.x?) UnlessI have bootrom 3.2.2.0019. Is that what people running thelatest have? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Thursday, April 12, 2007 00:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 501 issue with latest firmware: sluggish keys It has nothing to do with actually dialing. Even trying to press end call or the speakerphone button does not work at times. Have tried removing side cars etc, but definately seems to be a bug in the 2.x code stream. On Apr 11, 2007, at 5:37 PM, Eric ManxPower Wieling wrote: Jim King wrote: I've seen an issue like this from time to time on 601s, even with the latest firmware. Not just the softkeys, but also the dial keys. The phones seem to run slow sometimes, failing to respond to a key press right away but getting to it eventually. It usually clears up after a few seconds. Also, I've noticed that the 601s sometimes ignore key presses altogether, just as you describe. I have not yet found a solution for this problem... Try setting this in sip.cfg: dialplan.impossibleMatchHandling=1 I suspect it is either 0 or 2 now. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.17 and Cisco 7940/SIP: bug or what?
Hi. I'm stuck into an odd situation. Here's what happens: 4 Thomson ST2030S 2 Cisco 7912 3 Cisco 7940 2 AAstra 480i Asterisk 1.2.17 Diva 4BRI + chan_capi I've just upgraded yesterday from Asterisk 1.2.13 to 1.2.17. Until yesterday, everything was just fine with 1.2.13. Immediately after the upgrade, *all* the 7940 are no more able to make calls, just receive them, while 7912 models as well as any other phone work fine. Firmware on 7940 is 8.6 (the latest one). The configuration for asterisk is really simple. After many hours guessing and reloading configuration changes, I've traced the full debug output from both asterisk logger and one 7940. Here's what happens 1) I dial the number on the 7940 (which, by the way is regularly registered as a peer and REACHABLE by asterisk) 2) the 7940 sends an INVITE to asterisk 3) Asterisk sends back a 407 Authorization required 4) The 7940 sends back an ACK 5) The 7940 sends a new INVITE which includes the MD5 challenge response 6) nothing happens in asterisk (nothing logged, even with full debug enabled) 7) the 7940 retries sending the INVITE many times, until it times out 8) I hang up the handset What on Earth is happening Why is not Asterisk logging the subsequent INVITEs from the phone? (BTW, these sip packets are logged by iptables, I just wanted to make sure they were received on the asterisk ethernet interface) ## Here's an extract from asterisk log: ## smtp-ms*CLI -- SIP read from 10.0.10.136:50393: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.10.136:5060;branch=z9hG4bK0cc1ada5 From: Cisco 7940 sip:[EMAIL PROTECTED];tag=0013c3677fdf00ae6752cb07-7fbc304d To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 Date: Thu, 12 Apr 2007 13:39:56 GMT CSeq: 101 INVITE User-Agent: Cisco-CP7940G/8.0 Contact: sip:[EMAIL PROTECTED]:5060;transport=udp Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: Cisco 7940 sip:[EMAIL PROTECTED];party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Content-Length: 274 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 19361 0 IN IP4 10.0.10.136 s=SIP Call t=0 0 m=audio 16946 RTP/AVP 8 0 18 101 c=IN IP4 10.0.10.136 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 0: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 (43) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.10.136:5060;branch=z9hG4bK0cc1ada5 (56) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 2: From: Cisco 7940 sip:[EMAIL PROTECTED];tag=0013c3677fdf00ae6752cb07-7fbc304d (76) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 3: To: sip:[EMAIL PROTECTED];user=phone (34) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 4: Call-ID: [EMAIL PROTECTED] (56) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 5: Max-Forwards: 70 (16) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 6: Date: Thu, 12 Apr 2007 13:39:56 GMT (35) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 7: CSeq: 101 INVITE (16) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 8: User-Agent: Cisco-CP7940G/8.0 (29) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 9: Contact: sip:[EMAIL PROTECTED]:5060;transport=udp (49) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 10: Expires: 180 (12) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 11: Accept: application/sdp (23) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 12: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE (65) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 13: Remote-Party-ID: Cisco 7940 sip:[EMAIL PROTECTED];party=calling;id-type=subscriber;privacy=off;screen=yes (105) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 14: Supported: replaces,join,norefersub (35) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 15: Content-Length: 274 (19) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 16: Content-Type: application/sdp (29) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 17: Content-Disposition: session;handling=optional (46) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 18: (0) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3474 parse_request:
Re: [asterisk-users] Polycom 501 issue with latest firmware: sluggish keys
also I've seen that not having the correct version of sip.cfg and phone1.cfg could cause weird problems. Make sure you are using the ones that came with the firmware. Mike wrote: Exactly. It's a weird issue, and I can't imagine what the problem is, except maybe for bad phones (but then again, why would the phones be only bad with 2.x?) UnlessI have bootrom 3.2.2.0019. Is that what people running thelatest have? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Thursday, April 12, 2007 00:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 501 issue with latest firmware: sluggish keys It has nothing to do with actually dialing. Even trying to press end call or the speakerphone button does not work at times. Have tried removing side cars etc, but definately seems to be a bug in the 2.x code stream. On Apr 11, 2007, at 5:37 PM, Eric ManxPower Wieling wrote: Jim King wrote: I've seen an issue like this from time to time on 601s, even with the latest firmware. Not just the softkeys, but also the dial keys. The phones seem to run slow sometimes, failing to respond to a key press right away but getting to it eventually. It usually clears up after a few seconds. Also, I've noticed that the 601s sometimes ignore key presses altogether, just as you describe. I have not yet found a solution for this problem... Try setting this in sip.cfg: dialplan.impossibleMatchHandling=1 I suspect it is either 0 or 2 now. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nagios asterisk monitoring
Hi, Let me join all of you, interested in such monitoring tool. Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to install asterisk on redhat ?
Dovid B wrote: I wrote this ages ago. You may want to get more current software than the URL's that are listed. I just changed the version numbers before doing the script ;) -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF problem with inbound calls on Toll-Free number
Hi all, I have asterisk 1.2.13 and problem is about DTMF.When i have incoming call on Toll-Free number asterisk accept DTMF digits but dial only first in context. Per instance: When i press 1 it is OK,but when i try to dial extension 700 asterisk dial only first digit(1) and i receive from asterisk invalid extension 7 in context...Extensions 700 exists.It seems asterisk dial only first digit. When i dial ordinary(not Toll-Free)number everyting is OK. Please help. Regards! Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit. http://farechase.yahoo.com/promo-generic-14795097___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AgentCallBackLogin vs AddQueueMember
On Wed, 11 Apr 2007, Kevin P. Fleming wrote: Alan Ferrency wrote: This means that all queue activity is associated with a SIP channel in the logs, which is not acceptable. This is why we added the 'membername' argument to the AddQueueMember application, so that queue logs can reflect a logical name for the queue member, regardless of what channel/interface they logged in from. Okay, 'membername' does seem like it will solve several of our concerns. Thank you for pointing out this option and its intended use. 1. a person cannot be logged into more than one phone 2. only one person at a time can be logged into a phone For points #1 and #2, you are correct that this logic will have to be built. If all the rest of our functionality is taken care of, solving these might not be a problem. There is no reason for you to do _anything_ today, other than to start thinking about how you want to do it in the future when you decide to upgrade to Asterisk 1.6 and have to replace it... ... which is exactly what I've been trying to do with this thread. We have no plans to upgrade asterisk out of the 1.2 branch, because at this point the implementation costs would be far too high, as long as all we'd get out of it is downtime followed by status quo. (We're still using 1.2.3, because from what I've read, the combination of features we require has serious deadlock race conditions in newer versions of Asterisk. Let's just say, this is far from ideal.) ... but acting today like the functionality has been removed and that you are being forced to rearchitect your system seems a little bit extreme (in my opinion, of course). This is not the way I am acting. My intent with this thread is to: * learn enough about the new solution to know whether it will serve our needs or not * if not, try to push development in the direction we will need, _when_ the time comes that we must upgrade * show anyone who's listening our specific use case for AgentCallbackLogin, which may or may not have been considered My intent has not been to try to stop the deprecation of AgentCallbackLogin. When the time comes that we do decide to upgrade and reconfigure, I will need a high level of confidence that the solution I propose will serve our needs, and will provide value comparable to the cost of implementation. I can't achieve that by ignoring the situation until the last minute. From the example new solution and related documentation I have read previously, I did not come away with the impression that it did everything we needed it to. Your clarifications have helped on several points that I missed. So from my personal perspective, I consider this thread at least partially successful. It may be that more complete documentation would help mitigate this (perceived) problem. At least it would let you answer e-mails such as mine, which are likely far more common than you'd prefer, with a single rtfm:// URL. As it is now, we have a chicken-and-egg situation. Since no one is required to stop using AgentCallbackLogin, few have stopped using it. So, there are few examples in the wild of how to reimplement specific feature requirements. This lack of examples increases the migration cost away from AgentCallbackLogin, and the circle is closed. Thank you for your help, Alan Ferrency ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] video phones and call files
Hi All, I have 2 GXV-3000 phones. Working fine when I manually call the phones. However, if I use a call file to initiate my call to phone 1, then the dial plan calls the second phone only the second phone shows video not the first phone. How can I get video showing on the first phone also? Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP: number to names
Hi all, Is it possible to configure an extension number to dial a sip address? For example: exten = 101,1,Dial(SIP/sip:[EMAIL PROTECTED]) That way I can dial to a sip name using my Hardphone that is not able to dial using names just numbers. Thanks in advance. Ronaldo. (I hope putting my sip address soon here) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.17 and Cisco 7940/SIP: bug or what?
Alberto Pastore wrote: Firmware on 7940 is 8.6 (the latest one). I had the same issue. I ended up moving back to firmware P0S3-07-4-00 on the phone. I did a telnet into the phone, did a show register and shaw some very weird info. Normally, I would see: LINE REGISTRATION TABLE Proxy Registration: ENABLED, state: REGISTERED line APR state timer expires proxy:port --- - -- -- 1 111 REGISTERED 115 106 drdos.info:5060 2 111 REGISTERED 115 38 drdos.info:5060 3 ... NONE 0 0 undefined:0 4 ... NONE 0 0 undefined:0 5 ... NONE 0 0 undefined:0 6 ... NONE 0 0 undefined:0 1-BU .1x NONE 0 0 undefined:0 Note: APR is Authenticated, Provisioned, Registered But, under 8x firmware the timers would be some huge number and the state would be registering. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Measuring audio file legth
On Thu, 2007-04-12 at 13:28 +0200, Suity Zsolt wrote: Hi, I have to set call length to 3min, but before hangup have to warn caller. There are many IVRmenu and submenu options with different warning audio. I have to measure somehow the audio file length and subtract it from 3 minutes. I have not tried this, so I may be off - but do you really have to do this? The documentation I have indicates that if there is an extension T in the context, that extension is used at the absolute timeout. So, would: exten = T,1,play your warning message exten = T,n,Hangup work? HTH, -- Bob Smither [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Delay to start sip registration after asterisk restart
Hi, My asterisk was working fine but today my calls won't out of my asterisk box. Restarting asterisk i need to wait around 10 min to can run sip show registry command. If i try to run this command before, i receive a error like: no such command. Why this happen ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 501 issue with latest firmware: sluggishkeys - new info
I found *something*. I've gone into my CPU graph (on the phone, in status - diagnostic). Two phones, one running 1.6.7 and one running 2.1.0, both on the same Hub, with the same general configuration (different SIP registration, and each using it's version-specific sip.cfg file). The pre-2.x phone is running with CPU load approaching 0% (0%-7%). The 2.x phone has tons of spikes in the 100% range. What could be causing this? Where do I start looking? Mike _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Fuermann Sent: Thursday, April 12, 2007 10:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 501 issue with latest firmware: sluggishkeys also I've seen that not having the correct version of sip.cfg and phone1.cfg could cause weird problems. Make sure you are using the ones that came with the firmware. Mike wrote: Exactly. It's a weird issue, and I can't imagine what the problem is, except maybe for bad phones (but then again, why would the phones be only bad with 2.x?) UnlessI have bootrom 3.2.2.0019. Is that what people running thelatest have? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Thursday, April 12, 2007 00:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 501 issue with latest firmware: sluggish keys It has nothing to do with actually dialing. Even trying to press end call or the speakerphone button does not work at times. Have tried removing side cars etc, but definately seems to be a bug in the 2.x code stream. On Apr 11, 2007, at 5:37 PM, Eric ManxPower Wieling wrote: Jim King wrote: I've seen an issue like this from time to time on 601s, even with the latest firmware. Not just the softkeys, but also the dial keys. The phones seem to run slow sometimes, failing to respond to a key press right away but getting to it eventually. It usually clears up after a few seconds. Also, I've noticed that the 601s sometimes ignore key presses altogether, just as you describe. I have not yet found a solution for this problem... Try setting this in sip.cfg: dialplan.impossibleMatchHandling=1 I suspect it is either 0 or 2 now. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay to start sip registration after asterisk restart
Hi Frederico, I sometimes have the same problem tooI think the problem is related to VoIP providers registrations. Are you using VoIP services on your PBX? Thank you. Giorgio Incantalupo Frederico Madeira wrote: Hi, My asterisk was working fine but today my calls won't out of my asterisk box. Restarting asterisk i need to wait around 10 min to can run sip show registry command. If i try to run this command before, i receive a error like: no such command. Why this happen ? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC audio clipping
I'll be sending Digium support the info they requested later today. I hope it helps. Greg Siemon wrote: No luck yet. No response from Digium support so I guess that they are still waiting for the Zaptel test code. Greg -Original Message- From: Stephen Bosch [mailto:[EMAIL PROTECTED] Sent: Thursday, 12 April 2007 4:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HPEC audio clipping Greg Siemon wrote: Thanks for the helps Stephen. I was running non standard gains but setting regain and txgain to zero (then reloading chan_zap.so) does not help. I still get the broken audio, in fact sometimes I don't get any audio at all. In testing the server just froze a number of times and had to be rebooted via the power switch. I am using the latest Zaptel 1.2.16 files and the latest fxotune from the 1.4 release and I still see this issue. Very interested to get this working but without the HPEC my server is rock solid (only have to reboot it when I install kernel updates). I don't believe it is my system but am happy to do any testing others may suggest. Have you had any luck with this, Greg? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best External PRI Gateway?
I'm currently looking to interconnect my Asterisk PBX system with the PSTN via a digital PRI/T1. I know a multitude of options exist for internal PCI cards (Digium/Sangoma/Rhino), I was wondering if anyone has any experience or recommendations of external PRI media gateways that support SIP. So far I've found: VegaStream Vega 400 Audiocodes Mediant 2000 MediaTrix 1531 However they are all expensive (over 3,000). Does any one have any other suggestions or experience with the above products? Thanks, Jameson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 501 issue with latest firmware: sluggishkeys - new info
Mike wrote: I found *something*. I've gone into my CPU graph (on the phone, in status - diagnostic). Two phones, one running 1.6.7 and one running 2.1.0, both on the same Hub, with the same general configuration (different SIP registration, and each using it's version-specific sip.cfg file). The pre-2.x phone is running with CPU load approaching 0% (0%-7%). The 2.x phone has tons of spikes in the 100% range. What could be causing this? Where do I start looking? If you have CDP enabled, try turning it off if your network does not use CDP. You set this in the boot menu (same place you set the phone for DHCP or static, etc) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best External PRI Gateway?
May i ask why not internal? On 4/12/07, jameson asterisk [EMAIL PROTECTED] wrote: I'm currently looking to interconnect my Asterisk PBX system with the PSTN via a digital PRI/T1. I know a multitude of options exist for internal PCI cards (Digium/Sangoma/Rhino), I was wondering if anyone has any experience or recommendations of external PRI media gateways that support SIP. So far I've found: VegaStream Vega 400 Audiocodes Mediant 2000 MediaTrix 1531 However they are all expensive (over 3,000). Does any one have any other suggestions or experience with the above products? Thanks, Jameson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax Blast over IP?
Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? I use Asterisk now for my phone system. Thanks! Wiley E. Siler Director of Information Technology Education 2020 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] www.education2020.com http://www.education2020.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk (1.4) and hints/presence/BLF
Playing with hints/presence/BLF on asterisk I've made the following discoveries. 1. The wiki at http://www.voip-info.org/wiki/view/Asterisk+presence says: If you add incominglimit=1 to your peer in sip.conf, the SIP channel will notify you when that extension is busy. As incominglimit is obsolete you can use call-limit. Also you don't need to limit it to one, just having a call-limit at all works. (Tried with call-limit 20). What is the logic behind the linking of presence to call-limit? 2. A phone is only seen as busy if it's received an incoming call. Outgoing calls don't change the state. Why? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Installing codec g729 on Asterisk 1.2.1 on FreeBSD 6.0
Hi, I'm having problems installing codec g729 on my Asterisk that's running on FreeBSD 6.0 codec_g729a.so module loads ok, but the register utility doesn't seem to register the license key correctly, because when I issue show g729 under Asterisk's CLI it says that the command is invalid. It doesn't matter how many times I run the register utility, it allways says that the license key I enter is available for registration. Under the unsupported directory of digium's FTP, there are only files for FreeBSD 5.4 What should I do? Any ideas? Thank you all. Alejandro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best External PRI Gateway?
That's just the thing. There are manifold options, but they are all quite expensive. -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Blast over IP?
On Thu, 12 Apr 2007, Wiley Siler said something to this effect: Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? Asterisk can send faxes, if you make it interoperate with a few well-known open-source utilities and/or software packages, depending on what precisely you want to do: http://www.voip-info.org/wiki-Asterisk+fax -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [asterisk-users] Which SIP phones to buy?
Drew Gibson wrote: Stephen Bosch wrote: Stephen Bosch wrote: I need to buy some new phones for our own offices. I've used only Polycom phones until now, but I'd like to broaden my experience. I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I don't want to broaden my knowledge... ...because I like to stay dumb. Of course, that's not what I meant :) We have Cisco, Aastra 480i and Grandstream GXP2000 phones in house. I only recommend the Cisco phones to people I don't like, overpriced and far too much work. The Aastra 480i is a good quality phone, on par with Cisco and probably with Polycom (though I've never used them). Voice quality is good, phone feels robust. Config is well documented and contained in two text files (one global, one MAC specific). Good web interface on the phone. Aastra support have been very responsive. Grandstream phones are lower quality but good value for money. Sound and feel of phones is not so good as Aastra or Cisco. Configuration is through a binary file, a bit fiddly, but quite manageable with a few scripts. Good web interface on the phone. Grandstream support have also been very responsive. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com I have Cisco 7960G, Polycom SP 501, Aastra 480i, Grandstream GXP2000 and Budgetone 200 desk phones in my test lab. Overall, I like the Cisco best. I even bought one for home use. Configuration was no more difficult than any other. The Cisco, Aastra and Polycom have similar voice quality. They're all very good handsets and speakerphones. Of these three, the Aastra is the only backlit display, but it is hard to read from an angle and the backlight is not very effective. Aastra is also very vulnerable to glare. The Cisco and Polycom are easier to read unless you are in a darkened room. The Grandstream GXP2000 and Budgetone 200 have nice, bright and easy to read displays, but the phone aesthetics are not up to par with the others. For daily use, the Cisco and Polycom buttons are smoothest. The Aastra is close, but not as comfortable to use. It seems that round buttons function better. The Grandstream buttons are just heavy and cumbersome. The Polycom is the biggest pain in the ass to initially configure because of the extended boot time. All other brands I've used boot within a minute and are ready to use. The Polycom takes around 4 and if you are using the web interface for initial configuration, you need to reboot frequently. Once you've worked out your configuration, new phone installs are pretty simple with any brand. The Aastra and Grandstream web interfaces are easy to use and you may make multiple changes and then reboot when you're done. The Cisco has no web interface. For routine provisioning, Cisco only supports tftp and telnet. The Polycom supports tftp, ftp, sftp, http and https. The Aastra supports tftp, ftp http. Placing a logo on the Cisco display is trivial. I have not been successful with any other brand so far. For PoE use, the Polycom and Aastra use 802.3af. Up to the 7970, Cisco used a proprietary PoE pin configuration and require a special cable to use with a standards compliant PoE switch. The cable is easy to make, but you have to ensure that users are aware of the difference. As for price, Drew is right about the high cost of Cisco. If I hadn't found one on eBay, my personal phone would likely be Aastra. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Measuring audio file legth
Bob Smither wrote: On Thu, 2007-04-12 at 13:28 +0200, Suity Zsolt wrote: Hi, I have to set call length to 3min, but before hangup have to warn caller. There are many IVRmenu and submenu options with different warning audio. I have to measure somehow the audio file length and subtract it from 3 minutes. I have not tried this, so I may be off - but do you really have to do this? The documentation I have indicates that if there is an extension T in the context, that extension is used at the absolute timeout. So, would: exten = T,1,play your warning message exten = T,n,Hangup What if he wants to warn the caller with 30 seconds remaining? Then 15? Then 5? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 301 questions
Lee Jenkins wrote: Stephen Bosch wrote: Sidetone can be set in the phone configuration; before you do that, though, I need to know what you mean by feedback. Sorry, should have been more detailed. It's a sort of background humming noise, almost like that if you placed the phone next to a high output electric device. Didn't hear it with Grandstream phone so I thought it may be something I can change in Polycom. Of course, it could also be that polycom is better phone, but more apt to pick up background noise that It's only on zap calls. I'm still playing with rxgain and txgain in hopes of resolving this. Basically, the humming noise increase with volume. What were your zapata gain settings before you started tweaking them? As far as gain goes -- what about the Polycom phone? Are you using the defaults from the SIP firmware package? Do you have other analog extensions connected to the line? I had a situation in which there was audible humming because there was a fax machine connected to the analog line between the Asterisk server and the demarcation point. Where is the Asterisk server located? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which SIP phones to buy?
Drew Gibson wrote: We have Cisco, Aastra 480i and Grandstream GXP2000 phones in house. I only recommend the Cisco phones to people I don't like, overpriced and far too much work. The Aastra 480i is a good quality phone, on par with Cisco and probably with Polycom (though I've never used them). Voice quality is good, phone feels robust. Config is well documented and contained in two text files (one global, one MAC specific). Good web interface on the phone. Aastra support have been very responsive. Grandstream phones are lower quality but good value for money. Sound and feel of phones is not so good as Aastra or Cisco. Configuration is through a binary file, a bit fiddly, but quite manageable with a few scripts. Good web interface on the phone. Grandstream support have also been very responsive. Thanks for the comments. I think I might give one or two Aastra sets a try, just for tire-kicking. Cheers, -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Measuring audio file legth
On Thu, 2007-04-12 at 13:28 +0200, Suity Zsolt wrote:
Hi,
I have to set call length to 3min, but before hangup have to warn
caller. There are many IVRmenu and submenu options with different
warning audio.
I have to measure somehow the audio file length and subtract it from 3
minutes.
I have not tried this, so I may be off - but do you really have to do
this? The documentation I have indicates that if there is an extension
T in the context, that extension is used at the absolute timeout. So,
would:
exten = T,1,play your warning message
exten = T,n,Hangup
work?
Yes, it works and I use it already! But when caller go to extension 1
there are a 10 second warning message and when go to extension 2 there are
a 6 second message and so on...
I should not to exceed 3minutes time limit, but use connection to last the
last seconds...
Because all the lines are same I want to make a macro
or something like this
exten = _X,1,Set(_AudioLength=MeasureSomehow(warning${EXTEN})
exten = _X,n,Set(TIMEOUT(absolute)=${${MaxCallDuration}-${AudioLength}})
exten = _X,n,DoSomeJob
exten = T,1,Playback(warning${EXTEN})
exten = T,n,Hangup
I can't figure out what command can I write instead of MeasurSomehow.
--
Suich
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Re: [asterisk-users] Best External PRI Gateway?
On Thu, Apr 12, 2007 at 11:59:00AM -0400, jameson asterisk wrote: I'm currently looking to interconnect my Asterisk PBX system with the PSTN via a digital PRI/T1. I know a multitude of options exist for internal PCI cards (Digium/Sangoma/Rhino), I was wondering if anyone has any experience or recommendations of external PRI media gateways that support SIP. So far I've found: VegaStream Vega 400 Audiocodes Mediant 2000 MediaTrix 1531 Also have a look at Patton SmartNode 4960 range. They are available in various configurations/numbers of channels, some of which are upgradable to more channels at a later date: http://www.patton.com/products/pe_printable.asp?category=354 We have the ISDN2 and Analogue versions of these gateways (same software) and so far they have been very reliable, and can be configured in a variety of fail-over situations in case asterisk or the connection to the server dies, incoming calls can be automatically routed either back out on another ISDN channel or out to another SIP/analogue gateway etc. Rob -- Robert Lister - London Internet Exchange- http://www.linx.net/ [EMAIL PROTECTED] - tel: +44 (0)20 7645 3510- RL786-RIPE ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Blast over IP?
Wiley Siler wrote: Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? No, but I can recommend that you read this to see why you shouldn't bother: http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.17 and Cisco 7940/SIP: bug or what?
Doug Lytle ha scritto: Alberto Pastore wrote: Firmware on 7940 is 8.6 (the latest one). I had the same issue. I ended up moving back to firmware P0S3-07-4-00 on the phone. I did a telnet into the phone, did a show register and shaw some very weird info. Normally, I would see: ... But why does 8.6 seem to work with previous asterisk 1.2.13?? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax Blast over IP?
Basically, I want to send bulk faxes to a list of my clients. It is time consuming for a person to individually fax so a blast type solution seems best. Over IP is of course to save money... Thanks for the link, reading now... Any suggestions for the blast then? Wiley E. Siler Director of Information Technology Education 2020 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:[EMAIL PROTECTED] www.education2020.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Thursday, April 12, 2007 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax Blast over IP? On Thu, 12 Apr 2007, Wiley Siler said something to this effect: Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? Asterisk can send faxes, if you make it interoperate with a few well-known open-source utilities and/or software packages, depending on what precisely you want to do: http://www.voip-info.org/wiki-Asterisk+fax -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with Sipura SPA 3000
Hello Francis,
I also hev asterisk and sipura. Can we chat online on gmail/yahoo. Let's
make some experiments... I hev the same problem like you.
On 4/12/07, Francis Augusto Medeiros [EMAIL PROTECTED] wrote:
On 10 de abr de 2007, at 23:05, James Harper wrote:
2 - How can I gain full control to the FXS? I mean, a simple * dialed
is
not sent for asterisk (the server) interpretation, probably because
it's
used by Sipura's suplementary services, I don't know. Also, is it
possible
to get a dial tone from ASterisk, instead of Sipura's? My goal with
this
is to provide users with direct access to the PSTN line pressing 0,
instead of collecting calls and making the call themselves, or at
least
making ignorepat to work!
A dialplan of '(S0:s)' will get your phone to jump straight into the
's' extension in asterisk as soon as someone picks it up. From
there you
can do something like:
It worked perfectly! Thanks!
[sip_ata_incoming]
exten = s,1,Answer
exten = s,n,DISA(no-password|sip_extension_in)
so Asterisk will give you dialtone and do the dialplan stuff for you.
From the 'sip_extension_in' context you can make a single '0' or '*'
call the PSTN line.
On the sip_extension_in, I entered the following
exten = 0,1,NoOp(outgoing call via POTS line to No. ${EXTEN:1})
exten = 0,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],120)
exten = 0,3,Congestion()
exten = 0,4,Hangup
However, when I press the 0, it does gives me a dialtone, but it
doesn't seem to be delivering the tones imediately. I even suspect it
isn't my PSTN tone after the 0. Is there something else?
Cheers,
Francis
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Re: [asterisk-users] Which SIP phones to buy?
The Cisco phones are quite good. The thing that most people don't tend to appreciate about them is that they all are designed essentially for mass-provisioning in large environments, and to operate with Call Manager. Provisioning them using their GUI/configuration interface on a one-off basis is a pain in the butt, this is absolutely true. They were never really intended to be used in that manner. If you can take the time to do the TFTP thing, though, they really are very wonderful, featureful, reliable, comfortable, and, I would go so far as to say, turn-key. Just my $.02. No interest in religious debate. -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which SIP phones to buy?
From overall apprecation feedback : #1 Polycom (Any) #2 Aastra 480i #3 Cisco 7940+ #4 Linksys SPA-94x On 4/11/07, Stephen Bosch [EMAIL PROTECTED] wrote: I need to buy some new phones for our own offices. I've used only Polycom phones until now, but I'd like to broaden my experience. I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I don't want to broaden my knowledge. Advice, anyone? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 501 issue with latest firmware: sluggishkeys - new info
Mike, Got off the phone with Polycom on this I have the same problem with my new 601 phone here (haven't seen the problem on the 650). I'm trying to find answers and Polycom's only got one reported case of this (which I find bazaar, but whatever). The problem was resolved, the problem was the user was using 1.6.5 configuration files with the 2.x firmware. Once the user put the config files that are sent with the firmware, problem went away. I cannot stress this enough, as minuet or insignificant a change may appear in the configuration files, or as similar as they look, use the ones provided with the firmware! Now, I'm not sure if we've done this ourselves, but I'm having one of our support guys today looking at it and getting new configuration files into our phones. I'll see if that resolves the problem or not. I may not have an answer back until Friday or Monday, but if any of you guys experiencing this issue want to try as well, be my guess. Mentioning this now only because this is the information that came from Polycom support as a resolved problem. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http:// voipstore.atacomm.com/ On Apr 12, 2007, at 10:37 AM, Mike wrote: I found *something*. I've gone into my CPU graph (on the phone, in status - diagnostic). Two phones, one running 1.6.7 and one running 2.1.0, both on the same Hub, with the same general configuration (different SIP registration, and each using it's version-specific sip.cfg file). The pre-2.x phone is running with CPU load approaching 0% (0%-7%). The 2.x phone has tons of spikes in the 100% range. What could be causing this? Where do I start looking? Mike From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Jason Fuermann Sent: Thursday, April 12, 2007 10:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 501 issue with latest firmware: sluggishkeys also I've seen that not having the correct version of sip.cfg and phone1.cfg could cause weird problems. Make sure you are using the ones that came with the firmware. Mike wrote: Exactly. It's a weird issue, and I can't imagine what the problem is, except maybe for bad phones (but then again, why would the phones be only bad with 2.x?) UnlessI have bootrom 3.2.2.0019. Is that what people running thelatest have? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Thursday, April 12, 2007 00:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 501 issue with latest firmware: sluggish keys It has nothing to do with actually dialing. Even trying to press end call or the speakerphone button does not work at times. Have tried removing side cars etc, but definately seems to be a bug in the 2.x code stream. On Apr 11, 2007, at 5:37 PM, Eric ManxPower Wieling wrote: Jim King wrote: I've seen an issue like this from time to time on 601s, even with the latest firmware. Not just the softkeys, but also the dial keys. The phones seem to run slow sometimes, failing to respond to a key press right away but getting to it eventually. It usually clears up after a few seconds. Also, I've noticed that the 601s sometimes ignore key presses altogether, just as you describe. I have not yet found a solution for this problem... Try setting this in sip.cfg: dialplan.impossibleMatchHandling=1 I suspect it is either 0 or 2 now. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 301 questions
Stephen Bosch wrote: Lee Jenkins wrote: Stephen Bosch wrote: Sidetone can be set in the phone configuration; before you do that, though, I need to know what you mean by feedback. Sorry, should have been more detailed. It's a sort of background humming noise, almost like that if you placed the phone next to a high output electric device. Didn't hear it with Grandstream phone so I thought it may be something I can change in Polycom. Of course, it could also be that polycom is better phone, but more apt to pick up background noise that It's only on zap calls. I'm still playing with rxgain and txgain in hopes of resolving this. Basically, the humming noise increase with volume. What were your zapata gain settings before you started tweaking them? .5/.5 As far as gain goes -- what about the Polycom phone? Are you using the defaults from the SIP firmware package? Not quite sure as I'm still getting familiar with the phone itself. I would say yes because I did not change any of those settings except to specify priority for codecs and to populate Line 1 variables. Do you have other analog extensions connected to the line? I had a situation in which there was audible humming because there was a fax machine connected to the analog line between the Asterisk server and the demarcation point. No, the line goes directly from the phone company terminated jack to the the Asterisk box. I also tried wedging a line filter between the Asterisk box and the jack, but no effect. Where is the Asterisk server located? It's located on a shelf in my office. The only other electronics that are close to it is the UPS for it, about 2 feet away. I' tried playing with the rxgain and txgain again and it sounds like its minimalized, but I can still hear it some although it is very low. It is lowest when the call is bridged. Highest after dial, but before bridged. Thanks again, -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom - Static IP
Noah, I am just using a dlink router for dhcp. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Wednesday, April 11, 2007 2:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom - Static IP Hi Steve - Is there a way to config a static ip address on a Polycom phone remotely ie. From a config file or a web browser? If you have a good DHCP server, you can use it to assign a static address to the phone's MAC. What DHCP server are you using? - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source VoIP client (on a webpage)
Dear Jason, Here in my company we use an applet it java IAX, and it functions very well! If to want to visit the URL is http://www.virgos.com.br, calls the service as 0800Web. Leonardo Silva 2007/4/5, Jason Wolfe [EMAIL PROTECTED]: I need to decide on the best way to add a voip SIP or IAX client to a website. I'm thinking that I'd like it to be inline, like an aplet, on the page. I've got some asterisk servers running to connect up to, so the real challenge is finding an easily integrated open source client. Any suggestions from those who know? Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Leonardo Silva fone: 16 8143-1146 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC audio clipping
Eric ManxPower Wieling wrote: I'll be sending Digium support the info they requested later today. I hope it helps. We have a developer working on extending Zaptel to support pre-echo audio capture right now, so that we can work on debugging these issues with real data instead of just conjectures :-) Stay tuned, a patch should be available for testing in the very near future. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Destar web interface problem
People, I have a Debian box with Asterisk and I've installed the Destar package in order to get web managing of my voip system. After I installed Destar, it runs on localhost:8080, but my server does not have X-Window to access to it so I can engter the web interface.. So how can I change localhost:8080 to IP_ASTERISK:8080 in order to access Destar via web from another PC ??? Really thanks, Alejandro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-Java website
Does anyone know who maintains the Asterisk-java web site at asterisk-java.org? The site seems to have been unavailable for a couple of days now. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax Blast over IP?
Either analog modems or a PRI, and Hylafax for automation, no VOIP involved there. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Thursday, April 12, 2007 10:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Fax Blast over IP? Basically, I want to send bulk faxes to a list of my clients. It is time consuming for a person to individually fax so a blast type solution seems best. Over IP is of course to save money... Thanks for the link, reading now... Any suggestions for the blast then? Wiley E. Siler Director of Information Technology Education 2020 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:[EMAIL PROTECTED] www.education2020.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Thursday, April 12, 2007 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax Blast over IP? On Thu, 12 Apr 2007, Wiley Siler said something to this effect: Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? Asterisk can send faxes, if you make it interoperate with a few well-known open-source utilities and/or software packages, depending on what precisely you want to do: http://www.voip-info.org/wiki-Asterisk+fax -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Blast over IP?
Wiley Siler wrote: Basically, I want to send bulk faxes to a list of my clients. It is time consuming for a person to individually fax so a blast type solution seems best. Over IP is of course to save money... Thanks for the link, reading now... Any suggestions for the blast then? My suggestions are in the reading material. Basically it boils down to you not using VoIP for fax. Lee. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Thursday, April 12, 2007 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax Blast over IP? On Thu, 12 Apr 2007, Wiley Siler said something to this effect: Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? Asterisk can send faxes, if you make it interoperate with a few well-known open-source utilities and/or software packages, depending on what precisely you want to do: http://www.voip-info.org/wiki-Asterisk+fax -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.17 and Cisco 7940/SIP: bug or what?
Alberto Pastore wrote: But why does 8.6 seem to work with previous asterisk 1.2.13?? That I wouldn't be able to answer. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] speex codec: Out of Buffer space
Hi, When I tried to use speex (8 khz) codec I got following warning messages on the Asterisk console. The other end was pjsip and I was testing this in local network. Here is a exact message: WARNING[6055]: codec_speex.c:237 speextolin_framein: Out of buffer space Has anybody had success in using Speex (8 khz) with Asterisk? My real interest is to use Speex (16 khz) but it seems like 16 khz speex is not supported with Asterisk. Any comments? Thanks, Madhuri No need to miss a message. Get email on-the-go with Yahoo! Mail for Mobile. Get started. http://mobile.yahoo.com/mail ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 501 issue with latest firmware: sluggishkeys - new info
Jessee J Holmes wrote: Got off the phone with Polycom on this I have the same problem with my new 601 phone here (haven't seen the problem on the 650). I am using an IP650 with the latest firmware (and the corresponding sip.cfg file) and I have seen this behavior. It is most noticeable when on-hook dialing, where I will dial two or three digits and then press the fourth digit and nothing appears on the display for 1-2 seconds for that keypress. Very odd indeed. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which SIP phones to buy?
Drew Gibson wrote: The Aastra 480i is a good quality phone, on par with Cisco and probably with Polycom (though I've never used them). Voice quality is good, phone feels robust. Config is well documented and contained in two text files (one global, one MAC specific). Good web interface on the phone. Aastra support have been very responsive. I've programmed who knows how many phones so far, so let me focus on this year... Cisco ... Overrated. I have one on my desk right now specifically being used for me to test security on it (writing an exploit against it to be specific). I don't even bother using it... Aastra 480i's. I can take a picture of my desk for verification for those who'd want it of two I have sitting here collecting dust. They're horribly documented. Their web interface is full of errors (Username/CallerID/Auth) of which unless you're used to doing it you will have issues programming these. Polycoms... The bane of my existence. If you plan on doing NAT, passing through Netscreens, Sonicwalls, etc., and you don't mind miserably wasting time, then these are for you! If you're a glutton for XML nonsense, waiting 2 minutes for a reboot after EVERY SINGLE CHANGE. This phone is for you! If you don't mind explaining the Americans with Disabilities ACT and how Polycom is the only vendor resetting volumes then this is for you! And yes I am aware I could make that static via xml so please don't bother with a but you can fix that this way... response. Snom, although not the greatest, within the past year I've had to deal with well over I would guesstimate 200 or so. Easiest to deal with. Grandstream... Sorry, there is only so much garbage I'm willing to keep around my desk. GXP 2000? Fisher Price toy looking phone I wouldn't bother with. Robert Greene wrote: I have Cisco 7960G, Polycom SP 501, Aastra 480i, Grandstream GXP2000 and Budgetone 200 desk phones in my test lab. Overall, I like the Cisco best. I even bought one for home use. Configuration was no more difficult than any other. This is what is within two feet of me right now. 2 Cisco 7960's, 3 Polycruds, 2 Aasta 480i's, Welltech piece of garbage, 1 Snom 360, 2 320's, unlimited 190's. Guess which one I used on a daily basis... Snom 360. @Home ... http://www.infiltrated.net/Mar2520074.jpg I have about 3 7960's for my CCVP lab studies... I have a Hitachi WiFi and a Snom 320. Guess which I use most... Hitachi so I could walk around, followed by Snom. I don't even want to bother with the Cisco phones. The Cisco, Aastra and Polycom have similar voice quality. They're all very good handsets and speakerphones. The 480iCT is questionable. The base is alright, nothing to boast about, the handset... Depends on the environment. As for price, Drew is right about the high cost of Cisco. If I hadn't found one on eBay, my personal phone would likely be Aastra. I could care less about pricing. I'd be more concerned with quality and ease of use for both the admin, and the user. Cisco would rank low on my list, so here goes mine in order... 1) Snom 2) none! (they're all pretty much the same to me) 3) none! (they all have their pros and cons) 4) Cisco 5) ASStra 6) Polycrud -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Catch all undefined numbers to play a nice message and restart
Hi there list!
I want to catch all numbers that don't exist, play a
nice message and restart operator, this is different
from dial i because that is for incorrect extensions,
an undefined number will give a busy signal, something
I don't like
You can search for the word irc to see my comments,
the line above is my latest unsuccessful test, thanks!
;
;
;
;
;
; begin extensions
;
;
;
;
;
;
[general] ;
language=es
; autofallthrough=yes
clearglobalvars=no
[globals]
; Definiendo variables para usarlas a traves de todo
el
; MINOMBRE=mailinator.net
; MITELEFONOFXO=
; OPERADORA=
;
; Si static esta en no, u omitido, entonces pbx_config
va a sobreescribir
; a este archivo cuando se cambien las extensiones.
Recuerda que todos los
; comentarios de este archivo desapareceran si pasa
eso.
;
; XXX Todavia no ha sido implementado XXX
;
static=yes
;
;
; si stati=yes y writeprotect=no, tambien puedes
guardar al dialplan con
; linea de comandos ejecutando 'save dialplan' y
borrando estos comentarios
;
writeprotect=yes
CONSOLE=Zap/1 ; pendiente entender *
TRUNK=Zap/1 ; Trunk interface *
TRUNKMSD=1 ; MSD digits to strip (usually
1 or 0) *
;
; Trunks
;[context] ;exten =
someexten,priority[+offset][(alias)],application(arg1,arg2,...)
[trunkint] ; International long distance
through trunk
exten = _9001.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld] ; Long distance context
accessed through trunk
exten =
_901ZX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunklocal]; Local eight-digit dialing
accessed through trunk interface
exten =
_9ZXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) ;
llamada local comun y corriente
exten = _90ZXS0,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
; 020, etc
exten = _9066,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) ;
066, etc
[trunktollfree] ; Long distance context
accessed through trunk interface
exten =
_901800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkpaypercall] ; Dangerous pay-per call!
exten =
_901900.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkcelular] ; Long distance context
accessed through trunk interface
exten =
_9044ZZ,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten =
_9045ZZ,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
;
; Contexts
[international] ; Master context for
international long distance
ignorepat = 9
include = longdistance
include = trunkint
[longdistance] ; Master context for long
distance
ignorepat = 9
include = local
include = trunkld
include = trunktollfree
include = trunkpaypercall
[mercadotecnia]
ignorepat = 9
include = local
[local] ; Master context for local,
toll-free, and iaxtel calls only
ignorepat = 9
include = default
include = parkedcalls
include = trunklocal
[record]
exten = s,1,Answer
exten = s,2,Read(RECORD|enter4digits|4)
exten = s,3,Playback(record-instructions)
exten =
s,4,Record(/var/lib/asterisk/sounds/recording/s-${RECORD}|wav)
exten = s,5,Wait(2)
exten =
s,6,Playback(/var/lib/asterisk/sounds/recording/s-${RECORD})
exten = s,7,ResponseTimeout(10)
exten =
s,8,Background(1toaccept2torerecord3torecordanother)
exten = 1,1,Hangup
exten = 2,1,Goto(s,3)
exten = 3,1,Goto(s,2)
[macro-stdexten];
;
; Macro de extensiones estandard:
; ${ARG1} - Extension (Pudimos haver usado
${MACRO_EXTEN} tambien aqui
; ${ARG2} - Aparato(s) a marcar
;
exten = s,1,Dial(${ARG2},20) ; Ring the interface,
20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1); Jump based on
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(${ARG1},u) ; If
unavailable, send to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they
press #, return to start
exten = s-BUSY,1,Voicemail(${ARG1},b) ; If busy,
send to voicemail w/ busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they press
#, return to start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything
else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press
*, send the user into VoicemailMain
[macro-stdexten-viejo] ; Standard extension macro:
; ARG1 es el numero de la extension
; ARG2 es sip al cual voy a marcar
exten = s,1,Dial(${ARG2},20,rt) ; Ring the interface,
20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If
unavailable, send to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If
they press #, return to start
exten = s-BUSY,1,Voicemail(b${ARG1}) ; If
Re: [asterisk-users] Asterisk-Java website
Hum, I know Stefan, he is an asterisk-java dev, but he is not online right now, I will let him know ASAP. Thanks! On 4/12/07, Doug Garstang [EMAIL PROTECTED] wrote: Does anyone know who maintains the Asterisk-java web site at asterisk-java.org? The site seems to have been unavailable for a couple of days now. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax Blast over IP?
Thanks all... Looks like I will have to let them know that FOIP is a no go and that we can automate on Asterisk though... Thanks! Wiley E. Siler Director of Information Technology Education 2020 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:[EMAIL PROTECTED] www.education2020.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Howard Sent: Thursday, April 12, 2007 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax Blast over IP? Wiley Siler wrote: Basically, I want to send bulk faxes to a list of my clients. It is time consuming for a person to individually fax so a blast type solution seems best. Over IP is of course to save money... Thanks for the link, reading now... Any suggestions for the blast then? My suggestions are in the reading material. Basically it boils down to you not using VoIP for fax. Lee. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Thursday, April 12, 2007 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax Blast over IP? On Thu, 12 Apr 2007, Wiley Siler said something to this effect: Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? Asterisk can send faxes, if you make it interoperate with a few well-known open-source utilities and/or software packages, depending on what precisely you want to do: http://www.voip-info.org/wiki-Asterisk+fax -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and hard phone configuration
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello Asterisk Gurus! I have a very simple question. I've just started playing around with Asterisk and BSD box. I also have grandstream ip phone and installed asterisk from ports. Now I'm on my very first steps to configure Asterisk. The question is: How do I make Asterisk communicate with my Grandstream hard phone? Thank you in advance. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGHovpUZGmaUWxLn8RAn9UAJ94exp6gs2PBWpMDiiNA69Mt78jhgCfYy71 eOq4eOuYi2uDpve+8YM2fp4= =+Jt7 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cacti/Nagios monitoring, what do you want graphed.
Hey guys, What are some of the numbers you guys want graphed? Anything that is a number, or any kind of information. Now I have Agents logged in and out # of queues total calls total channels What else? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and hard phone configuration
Hi, It's really a simple question! I've just started playing with asterisk too, and I think what you want could be found in the 4th chapter of Asterisk: The Future of the Internet. It's a open book you can download from http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11. I hope it'd helped you. Ronaldo. On 4/12/07, Ilya Vishnyakov [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello Asterisk Gurus! I have a very simple question. I've just started playing around with Asterisk and BSD box. I also have grandstream ip phone and installed asterisk from ports. Now I'm on my very first steps to configure Asterisk. The question is: How do I make Asterisk communicate with my Grandstream hard phone? Thank you in advance. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGHovpUZGmaUWxLn8RAn9UAJ94exp6gs2PBWpMDiiNA69Mt78jhgCfYy71 eOq4eOuYi2uDpve+8YM2fp4= =+Jt7 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cacti/Nagios monitoring, what do you want graphed.
On 4/12/07, Brandon Kruse [EMAIL PROTECTED] wrote: Hey guys, What are some of the numbers you guys want graphed? Curious how you are going to do this and will it be backwards portable. One of our engineers wrote an app that queries the manager interface to build RRD data. That's sent over to Cacti to monitor active calls on a box. I could think of many things queue related that would be good to have, but then again, shouldn't that be done somewhere else? Being able to break out calls in things like Zap(trunks), SIP, IAX, etc would be very useful. I have an 8 port T1 card and it would be nice to see how many calls I have on each port. Thanks for getting this going Brandon, I'll follow closely. We are heavy Nagios and Cacti users here. -Brian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sharing trunks between asterisk machines
Hello eveybody, I've been looking for a way to share trunks between two asterisk servers. I guest I have to use Dundi, but I've not found the exact method yet. I need a way to allow users registered in one server to use the another server's trunks in the case the first server's trunks were busy and vice versa. Is this possible? Thank you so much, any comment will be useful. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sharing trunks between asterisk machines
On Thu, 12 Apr 2007, Edgar Guadamuz said something to this effect: I've been looking for a way to share trunks between two asterisk servers. Provided that the Asterisk servers can be set up to hold identical SIP contacts (URIs), you can just set up a dialplan such that it fails over if a primary trunk is unavailable and sends the call via SIP to another server which can then put it on its trunks. -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Which SIP phones to buy?
I've had experience with quite a few different phones, so I think I'm qualified to drop my two cents: Alex is quite right that the Cisco phones are only designed to be used with Cisco Call Manager. They are capable of being decent SIP telephones, but Cisco won't provide the documentation so that you can use them effectively with anything other than Cisco Call Manager, so that's the deal killer. Like everything else Cisco, they're also ridiculously expensive. Despite what Alex says, the Cisco SIP phones have plenty of fundamental flaws. I have a number of expensive 7970G phones with a beautiful color display. Each of the various SIP firmware versions available for that product has a serious flaw. The most acceptable version is about a year old. It's biggest flaw is that the Message Waiting Indicator doesn't work. Most of the other SIP firmware versions won't register with Asterisk. If you are planning to usee Asterisk, save your money and your sanity and buy something else. In my last project, I used the Aastra 480i phones. Yes, the documentation is lacking, but that's largely because the platform was evolving quickly. Aastra has excellent and responsive technical support via e-mail. Finally, the customer was very satisfied with the quality and the price of the 480i phones. In my latest project, I used the newer Aastra 57i and 57i CT phones. It is obvious that these phones derived from the 480i software, but they are much faster and more full-featured with great displays, etc. The initial documentation with these is fairly good and complete. I have them doing all kinds of things, including using the XML capabilities to push server applications to the display, update the softkeys in real-time, etc. As contrasted against Cisco, Aastra even provides PHP include files to greatly simplify web development on whatever platform (Asterisk, Sylantro, etc.) you are using. The 57i phones are a little expensive, but they are a top-notch product that works very well with Asterisk right out of the box. Plus, they look and sound great and have 12 softkeys that shift to 20. One of the others that responded to your question mentioned something about setting up a TFTP server and I want to elaborate on that a little. If you are deploying more than a small handful of phones, you will want to setup a TFTP server anyway. It would be muy loco to try deploying and supporting a few dozen phones otherwise. Many of the phone's features aren't even accessible through the web interface anyway - you have to have a TFTP server and make use of the configuration files for full functionality. And that applies to Aastra, Cisco, Polycom or whatever. Finally, it can take a fair amount of labor to configure Asterisk and your particular phone to work together as a system. Don't kill yourself by attempting to mix and match various phones on the same system as that seriously increases the complexity. Keep it simple. For what it's worth Ken ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sharing trunks between asterisk machines
Another way is to run the calls through a SIP proxy such as SER which can hunt through two Asterisk UA endpoints, depending on a variety of parameters including failure at a primary and fallback to a secondary. -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Which SIP phones to buy?
Ken, You have certainly had experience with a broader range of phones, so I have no doubt you can lend more insight on this count. But for what it's worth, my experience is largely confined to the Cisco 7960s. I've never had any trouble getting any SIP firmware image to register with Asterisk, nor configuring them by hand against Asterisk in network situations that don't lend themselves to autoprovisioning setups. And I've never had any issues with features like MWI or various other notifications. About the only thing I've run into is that some of the older default dialplan.xml's tend to be hostile to numbers that start with 8xx, such as (but not exclusively) toll-free numbers. TFTP provisioning is, of course, the best way to blank those. Other than that, I've got no complaints. Awesome speakerphone, nice configuration interface, conferencing features, etc. And while they are obviously Call Manager-centric, I wouldn't go so far as to say that Cisco provides no documentation on how to get them to work otherwise; I've needed -- and found -- it. Thanks, -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] real time billing system
On Fri, 2006-09-29 at 11:12 -0500, Pato Valarezo wrote: Hi, sorry for the question, i've been searching for a real time billing system for asterisk with zap/sip support, for use in post paid systems like locutorios, do you know of or use any ? Give a try to StarshopOSS: http://www.starshop-online.com/howto/how_to_setup_voip_calls_in_your_cybercafe_with_starshop_3.htm Regards, thanks -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sharing trunks between asterisk machines
Thank you Alex and It would be possible to do that using IAX too,
wouldn't it?
I mean something like
exten=_9NXX,1,Dial(Zap/g0/${EXTEN:1})
exten=_9NXX,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
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[asterisk-users] zaptel/ssh interaction
I hope I don't get flamed the first time I post to a new list. I have spent a couple of hours poking around without seeing anything like this. The problem is, as soon as I load the Zaptel drivers (with a TDM-31B card), ssh into or out of the server is broken. Trying to ssh in, I get: RSA_public_decrypt failed: error:0407006A:rsa routines:RSA_padding_check_PKCS1_type_1:block type is not 01 key_verify failed for server_host_key If I try to ssh out, I get: hash mismatch key_verify failed for server_host_key This makes administering the server remotely impossible, so it's a fairly large problem for me right now. Anybody ever seen anything like this? It is easy to reproduce: modprobe zaptel and it's broken. modprobe -r zaptel and it works fine. Also, and probably somehow related, when the zaptel drivers are loaded, the sound through the sound card is screwed up. I can still hear it, but there are extraneous beeps, crackles and pops in it. Aside from this, I love my new asterisk system, and my wife has almost gotten used to having to dial 9 to get out of the house :-) --Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SCCP Firewall rules?
Has anyone tried to pass sccp through a cheap router / nat box? I have gotten sccp to go through a cisco pix just fine, but I can't seem to get it to go through a ipfilter box or a basic netgear / linksys router. I was under the impression that sccp was a lot more nat friendly, but at the moment I can get the phone to speak to the asterisk server, but not pass any audio either way. Thanks Shawn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Spandsp-0.0.3 and asterisk 1.2
Hi All Has anyone managed to get Asterisk 1.2 faxes working reliably with spandsp 0.0.3? I am running Asterisk 1.2.17 and spandsp 0.0.3pre28 with a Digium b410p card. Everything compiled smoothly but only about 70% of faxes come through ok. Debugging shows nothing more than: app_rxfax.c: Fax receive not successful - result (11) Unexpected message received. The files are only 8 bytes long??? Garth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RAGI channel_status() never returnes
Hi there,
I am new to this ML. Recently I started working on Asterisk 1.4 + RAGI +
Ruby on Rails to create a call history browser.
To record call history, I am trying to capture dialup, answer and hangup
events. To check what status a call is, I use channel_status() that RAGI
provides.
I am having a trouble on this function. In a polling loop that checks
call status, the first call of channel_status() returns -1 that
indicates a failure. Then the second call never returns to caller once
called. Below is a debug log and the code snippet of my app:
debug log:
[2007-04-13 08:42:40] INFO Dialup
[2007-04-13 08:42:40] INFO Dialing
[2007-04-13 08:42:40] INFO Waiting for call to be done
[2007-04-13 08:42:40] INFO -1
[2007-04-13 08:42:48] INFO Hangup
code:
require 'ragi/call_handler'
require thread
class CallHistoryHandler RAGI::CallHandler
def dialup
logger = WEBrick::Log::new
logger.info(Dialup)
hangup = false;
Thread.start {
# Keep running until the call is hung up
while hangup == false
# The 1st call returns -1
# The 2nd call never returns
status = channel_status()
# To show the polling loop keeps on running
logger.info(Waiting for call to be done)
if status == 4
logger.info(Ringing)# Still ringing
elsif status == 6
logger.info(Answered) # Call is answered
else
logger.info(status)
end
end
# dial() is done
logger.info(Call has been hung up)
}
logger.info(Dialing)
dial('IAX2/' + @params['agi_extension'])# Place a call
hangup = true; # Notify the call is hung up.
logger.info(Hangup)
end
end
Does anyone have clue about why the 2nd call of channel_status() never
returns?
Best regards,
Hisashi Adachi
[EMAIL PROTECTED]
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Re: [asterisk-users] Sharing trunks between asterisk machines
Certainly. Any signaling / trunking protocol will do, in principle.
On Thu, 12 Apr 2007, Edgar Guadamuz said something to this effect:
Thank you Alex and It would be possible to do that using IAX too,
wouldn't it?
I mean something like
exten=_9NXX,1,Dial(Zap/g0/${EXTEN:1})
exten=_9NXX,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
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--
Alex Balashov [EMAIL PROTECTED]
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[asterisk-users] Re: Which SIP phones...
Victor Hoodicoff wrote:
I think your impressions of Aastra are outdated. Install the latest
firmware, download the latest documentation and test and THEN give an
opinion!
Did you miss the part when I wrote I have Asstras sitting on my desk
collecting dust. I program on average about 5 per month, deal with
about 40+ per day. They're as impressive as that Hyundai in the lot
next to the Aston Martins.
I totally agree about the Cisco and Grandstream. People like to think the
Cisco are great because they paid too much for them. People want to like
the Grandstream because they are so cheap. I have no interest in dealing
with Polycoms unwillingness to support Asterisk.
Cisco is overrated. Grandstream... You get what you pay for. I don't know
who you deal with I have direct contacts with Polycoms to get what I want
when I need it.
Anyhow after doing a sip show peers and saving it to a file called UA
here is a summary from 2 servers I have to deal with on a daily basis...
This doesn't include others, these are my main two headaches...
The files UA are nothing more than more or less:
asterisk -rx sip show peers|awk -F / '{print $1}'|grep -v \.\|[a-z-]|uniq
UA
(Server 1 of 3. Each w/about 175-200 peers)
184 sip peers [139 online , 45 offline]
[EMAIL PROTECTED] ~]# grep -ic snom UA
93
[EMAIL PROTECTED] ~]# grep -ic polycom UA
43
[EMAIL PROTECTED] ~]# grep -ic grandstream UA
3
(Server 1 of 3. Each w/ 150-75 peers using DRBD between servers)
74 sip peers [164 online , 10 offline]
Verbosity is at least 10
xxx-1:~# grep -ic cisco UA
59
xxx-1:~# grep -ic poly UA
105
I can show another server with nothing but Snom's but I rarely
need to configure anything there...
Of these two PBX's... I get less calls about the Snoms with the
exception of Daylight Savings Time... Polycom is the most
problematic followed by Cisco... Aastra's... Most people that
buy those are usually a SoHo business who want maybe one or
two.
--
=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
echo @infiltrated|sed 's/^/sil/g;s/$/.net/g'
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
How a man plays the game shows something of his
character - how he loses shows all - Mr. Luckey
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[asterisk-users] Outside Network PAP and also Outside Network eyeBeam Soft Phone
I have been trying to setup a PAP2 adapter on a remote network but can't seem to get it to work. The unit will register with the server and it can make calls to extensions on the Asterisk server but it can't receive any calls and it can't make any calls outside of the Asterisk server. I also have a eyebeam soft phone that works when it is inside the network but when I am on other networks it won't. It will also register and it will make calls and will receive calls but there is no audio either way on it. Can someone point me in the right direction, or is there a Asterisk expert willing to look at it with me for pay of course? TIA, Andy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel/ssh interaction
Greg Woods wrote: I hope I don't get flamed the first time I post to a new list. I have spent a couple of hours poking around without seeing anything like this. The problem is, as soon as I load the Zaptel drivers (with a TDM-31B card), ssh into or out of the server is broken. Trying to ssh in, I get: RSA_public_decrypt failed: error:0407006A:rsa routines:RSA_padding_check_PKCS1_type_1:block type is not 01 key_verify failed for server_host_key If I try to ssh out, I get: hash mismatch key_verify failed for server_host_key This makes administering the server remotely impossible, so it's a fairly large problem for me right now. Anybody ever seen anything like this? It is easy to reproduce: modprobe zaptel and it's broken. modprobe -r zaptel and it works fine. Also, and probably somehow related, when the zaptel drivers are loaded, the sound through the sound card is screwed up. I can still hear it, but there are extraneous beeps, crackles and pops in it. Aside from this, I love my new asterisk system, and my wife has almost gotten used to having to dial 9 to get out of the house :-) I suspect IRQ Sharing. cat /proc/interrupts to see. If you are seeing more than 1 device on the same IRQ, see the mailing list archives on ways to resolve this. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
