Re: [asterisk-users] keep one line open
Check who is dialing this line by CallerID, if it is not your user - just drop the call. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of gilbert saunders Sent: Thursday, April 17, 2008 8:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] keep one line open hi i have multiple lines going to my asterisk box etc 0282549087 , 028 3659874 , 0285469658 etc. is it possible to keep users from using the 0282549087 line always open that it only allows a certain user to make outgoing calls on it? _ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try http://us.rd.yahoo.com/evt=51733/*http:/mobile.yahoo.com/;_ylt=Ahu06i62sR8H DtDypao8Wcj9tAcJ%20 it now. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] keep one line open
He said outgoing calls. Its simple. Just put it in a separate zap group, structure your dialplan (with AGIs or GotoIfs) so that only a particular user dials on it. On Thu, 2008-04-17 at 09:17 +0300, Mindaugas Kezys wrote: Check who is dialing this line by CallerID, if it is not your user – just drop the call. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO – Advanced Billing for Asterisk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of gilbert saunders Sent: Thursday, April 17, 2008 8:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] keep one line open hi i have multiple lines going to my asterisk box etc 0282549087 , 028 3659874 , 0285469658 etc. is it possible to keep users from using the 0282549087 line always open that it only allows a certain user to make outgoing calls on it? __ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.111.111.320 x200 www.emergen.biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP calls
Simon wrote: Hi There, We have our Asterisk box using a external SIP provider for outgoing calls over our DSL line. This seems to be going well... But i do have the ability to set some QOS ports in our linksystem DSL router... Its faily basic, so im wondering if it will help at all... We can specify High, Med, Low settings for: FTP, HTTP, Telnet, SMTP and POP3. Plus we have the ability to specify up to 3 ports for the same settings. Is this worth doing? If so, what ports should i specifiy? Simon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you can, try giving the highest priority to the UDP protocol or the provider IP address. Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] compilation of asterisk 1.4.19 with ilbc already on system
--- Kevin P. Fleming [EMAIL PROTECTED] wrote: Vieri wrote: So basically I'm wondering if the Asterisk make/configure process could do steps 1 and 2 automagically for me. I can't find any other Linux distribution that provides libilbc, so this would be a very Gentoo-specific change if we did it. Also, we'll have the iLBC source code back in the main distribution in the near future when the licensing issues are worked out, so for everyone else this will become a non-issue. I didn't know that. If that's the case then never mind my request. Do you see any particular advantage to using the system-provided libilbc, given that we use it in static (not shared object) form and it would have to be relinked into Asterisk if it got upgraded anyway? No real advantage but I had to patch Gentoo's Asterisk 1.4.19 ebuild (ie package) to avoid losing /usr/lib/asterisk/modules/codec_ilbc.so and I took advantage of the fact that there already exists an ilbc ebuild for Gentoo. If I want Gentoo users to upgrade seamlessly via the package management system I can think of only two choices: 1) change the 1.4.19 ebuild so that it runs the contrib/scripts/getilbcsource.sh before compiling (but I would still need to patch that script to remove the read) 2) change the * 1.4.19 ebuild so that it can take advantage of ilbc provided by another ebuild (via dependency). I proposed option 2 for Gentoo users and it's working fine for my system. However, if you plan to get the iLBC source code back in the main distribution soon then that's even better and as I wrote before, you can ignore my request. Thanks, Vieri Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup conundrum with RxFAX
On Wed, 16 Apr 2008, lordfuknowsyou wrote: My thoughts now are to actually do a hangup at the end of the RxFAX and rely on a 'h' extension to pick it up and carry on with the 2nd half (which is PDFing and emailling the fax), but I'm concerned I'm going to lose the channel variables as it suggests on the wiki, so I'll lose the REMOTESTATIONID string and caller ID... Hi. Thats what I do and have not had a problem, we only do maybe 10-20 faxes a week though. I set my channel variables in a macro and then goto a context receivefax where I enter on s,1,Rx.Fax , on hangup I do the actual mailing and sending of the fax. Before the sending though I make sure the fax actually exists. Thanks for this. I'll give it a go! Cheers, Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] keep incoming codec same as outcoming on sip proxy
Hi, i have two computer with asterisk. One is a SIP proxy that Dial() the other. It is possible to be sure that the proxy does not make transcoding in any case and Hangup() the call if the Second asterisk does not support the codec ? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] keep incoming codec same as outgoing on sip proxy
Hi, i have two computer with asterisk. One is a SIP proxy that Dial() the other. It is possible to be sure that the proxy does not make transcoding in any case and Hangup() the call if the Second asterisk does not support the codec ? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Warning 2512
Hi all, I have been seeing a lot of the following warning messages on my asterisk cli. Can naybody tell why these messages are showing up. I am using only SIP to make calls from m asterisk. [Apr 17 04:52:24] WARNING[2512]: chan_sip.c:6480 determine_firstline_parts: Bad request protocol Bad event Also it will be great if anybody can tell where i can find the explanation of all the warnig codes and error codes of asterisk if there is any. -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] imap voicemail
Hello. I'm trying to use gmail's imap feature w/ asterisk imap voicemail. I compiled c-client with the following settings: make lr5 IP6=4 and asterisk with: ./configure --with-imap=/usr/src/imap-2007a/ However if i enable any if the imap settings in voicemail.conf, asterisk starts acting funny and dosent allow any calls imapserver=imap.gmail.com imapport=993 mapfolder=Voicemail Where did I go wrong? -- Moshe Brevda, CTO ipconnect, ltd. 26 Strauss St., Jerusalem, Israel W. 1.800.800.456 (+9722.569.5295) M. +97254.666.1367 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Warning 2512
In article [EMAIL PROTECTED], Rizwan Hisham [EMAIL PROTECTED] wrote: I have been seeing a lot of the following warning messages on my asterisk cli. Can naybody tell why these messages are showing up. I am using only SIP to make calls from m asterisk. [Apr 17 04:52:24] WARNING[2512]: chan_sip.c:6480 determine_firstline_parts: Bad request protocol Bad event Also it will be great if anybody can tell where i can find the explanation of all the warnig codes and error codes of asterisk if there is any. The [2512] is not a warning code. It is just the process ID of the Asterisk process or thread that generated the warning. The next part of your message (chan_sip.c:6480) shows the source file and line number where the error was generated. You can go to that point in the file to see what kind of checks it was making. You can also turn on SIP debugging at the Asterisk CLI prompt to see the packets sent to/from Asterisk. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
Vieri wrote: Did you try a show channels to see if there were stale channels for peer 200? I had the same problem you describe but it was due to hung channels (used * 1.4.18.1 with rtp*timeout and saw inuse peers during the pre-timeout periods even though the agents weren't on a call). No, i don't , but how do do you fix this problem ? with rtp timeout ? Slds. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Warning 2512
I just saw the sip debug and its showing that for every notify request, asterisk is sending a bad request response. here is the debug --- SIP read from 70.80.000.00:1031 --- NOTIFY sip:69.90.111.11:9060 SIP/2.0 Via: SIP/2.0/UDP 70.80.000.00:1032;branch=z9hG4bK-ade549bd From: Blake sip:[EMAIL PROTECTED] [EMAIL PROTECTED] ;tag=90115683e082af23o0 To: sip:69.90.111.11 Call-ID: [EMAIL PROTECTED] CSeq: 7741 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA2102-5.2.3 Content-Length: 0 - --- (10 headers 0 lines) --- --- Transmitting (no NAT) to 70.80.000.00:1031 --- SIP/2.0 489 Bad event Via: SIP/2.0/UDP 70.80.000.00:1032;branch=z9hG4bK-ade549bd;received= 70.80.000.00 From: Blake sip:[EMAIL PROTECTED] [EMAIL PROTECTED] ;tag=90115683e082af23o0 To: sip:69.90.111.11;tag=as3ef6a439 Call-ID: [EMAIL PROTECTED] CSeq: 7741 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Why is it doing so? On Thu, Apr 17, 2008 at 2:36 AM, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Rizwan Hisham [EMAIL PROTECTED] wrote: I have been seeing a lot of the following warning messages on my asterisk cli. Can naybody tell why these messages are showing up. I am using only SIP to make calls from m asterisk. [Apr 17 04:52:24] WARNING[2512]: chan_sip.c:6480 determine_firstline_parts: Bad request protocol Bad event Also it will be great if anybody can tell where i can find the explanation of all the warnig codes and error codes of asterisk if there is any. The [2512] is not a warning code. It is just the process ID of the Asterisk process or thread that generated the warning. The next part of your message (chan_sip.c:6480) shows the source file and line number where the error was generated. You can go to that point in the file to see what kind of checks it was making. You can also turn on SIP debugging at the Asterisk CLI prompt to see the packets sent to/from Asterisk. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call forking feature
Hi, I have 2 wireless phones. I tried to register both the phones with the same number say 3000 to asterisk. But at any time i am able to see that only one phone is being registered. I want to test the call forking feature. How do I do this? Please help me in this regard. Thanks Regards, Jahnavi. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] buying cards from pakistan
Hi all, i want to buy a pci or whatever card for asterisk to plug in my telephone line into it and use asterisk as a pbx. i have only one telephone line at home. can you recommend me a simple cheap card which i can buy in pakistan. I live in pakistan, and i dont know any dealers here who sell asterisk cards. if someone knows where to buy cards in pakistan, plz tell me about it. -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Simon wrote: | Is this worth doing? If so, what ports should i specifiy? http://www.bricklin.com/qos.htm -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQIVAwUBSAcxA4OeOV2sx4+mAQKrKg//Vs5n0K1m1+C+sTSol2a1MbFHU/QCh1fT u5IwLOvQwcDvxOHikYYk8Ornm5FEbp4cewnCVKS9BeLkurlaQ1qXiwPdCHhnrNhn q2ufIskPudXn2cNj9pbAkZhmNL7R7m1XruITBGrXvV4d2WAZy6lmNGnVOhipU5ff HIV1aAacawCJG6oJjpD/NvjydHYwP6KgvOJkcLICrb7FEC4bfDfULQxThFF9Jzf3 aMzvddM+GdBGzhT0q7FH7JGQmuahlN1kdIyLY8Rw+/ouEgm4xYeyZ486JaBk2xOK 5ZnYqXXLmNuB7LPIkSCp2Fi5usFUBrKq8nanbvonw9Te3pILCG/FAfg/+O3Y2ZQb 0aZsUW13BHj9hfiZYLnKGCeJV1hLLuLWH+fP7E9kzFbi8ls7/Ke+oe7l8fgRFfzt oaInPjl1tshzbaOesSX1H8OI5QyfGgmuhyVu5E0tFmy9HX7QnxxBrI/GXMwQ3F6/ +Qv058sJ5qjQtGMi0fI6GoDa3xQRCzyBgWjuOBHhk64FVnMs3Rdoti69YhsWH52a LQMleyChhVQ0nrP5eVaykEryLNRw7jDV1X/ivtPNOHfQ0fN8337AHPmmKpzo22sK xdzK+RY1I/qZ+SOD6YPaiKjxaB9gqDPe7jGy41NlGsjnrgUjJh2c2/tvcyDYaW4u 0J5kiHsMXLI= =oY+k -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call forking feature
Asterisk only allows a single contact per SIP account so to do forking you'll need to use two SIP accounts and put them both in the Dial command. Or you could use OpenSER. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] imap voicemail
Hello. I'm trying to use gmail's imap feature w/ asterisk imap voicemail. I compiled c-client with the following settings: make lr5 IP6=4 and asterisk with: ./configure --with-imap=/usr/src/imap-2007a/ However if i enable any if the imap settings in voicemail.conf, asterisk starts acting funny and dosent allow any calls imapserver=imap.gmail.com imapport=993 mapfolder=Voicemail Where did I go wrong? Note that port 993 is IMAPS and I think you are telling it to use IMAP. Note also that Asterisk can get hung once in a while when using IMAP storage. Try netstat -an | grep 5060 and see whether you have a queue building there. __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
On Thu, 2008-04-17 at 07:16 -0400, sil wrote: Simon wrote: | Is this worth doing? If so, what ports should i specifiy? http://www.bricklin.com/qos.htm Yeah, well, that's all fine and dandy as long as more capacity is an option. Many people are already subscribed to the most capacity available to them and using it. b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Constant ''CHANUNAVAIL' on PRI for Outgoing Only
Hi, I'm hoping that somebody could possibly assist me with this. I've tried everything and I believe that my settings and configurations are 100% - CentOS 5.1 - 2.6.18-53.1.14.el5 Asterisk 1.4.19 libpri-1.4.3 zaptel-1.4.9 Connected via a Digium TE122P to a E1 PRI Incoming on any one of the numbers assigned to the E1 work fine and arrive at the Asterisk demo. No outgoing calls work and the following error is given: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/4700-085de988, Zap/g0/0215512345|300|Ttr) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g0/0215512345 -- Channel 0/1, span 1 got hangup, cause 44 -- Forcing restart of channel 0/1 on span 1 since channel reported in use -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/4700-085de988' status is 'CHANUNAVAIL' -- B-channel 0/1 successfully restarted on span 1 Here are all my settings related to this including an attached output of a pri intense debug span 1 as span-debug.txt I really hope that somebody is able to assist me as the Telco says nothing is wrong on their side (even though they are sending somebody out to come verify) Thanks in advance! /etc/zaptel.conf # # Zaptel Configuration File # span=1,1,0,ccs,hdb3,crc4 loadzone = za defaultzone= za bchan=1-15,17-31 dchan=16 #channel=1-15,17-31 /etc/asterisk/zapata.conf ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn signalling=fxs_ks rxwink=200 usecallerid=no hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=800 rxgain=0.0 txgain=-1.0 group=0 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busypattern=2500,500 busycount=2 callprogress=no hanguponpolarityswitch=no callerid=asreceived cidsignalling=v23 cidstart=ring ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown priindication = outofband callerid=asreceived jitterbuffers=6 ; PRI card - 1st span switchtype = euroisdn signalling = pri_cpe group = 0 context = demo channel = 1-15,17-31 [EMAIL PROTECTED] ~]# cat /proc/zaptel/1 Span 1: WCT1/0 Wildcard TE122 Card 0 (MASTER) HDB3/CCS/CRC4 IRQ misses: 22 1 WCT1/0/1 Clear (In use) 2 WCT1/0/2 Clear (In use) 3 WCT1/0/3 Clear (In use) 4 WCT1/0/4 Clear (In use) 5 WCT1/0/5 Clear (In use) 6 WCT1/0/6 Clear (In use) 7 WCT1/0/7 Clear (In use) 8 WCT1/0/8 Clear (In use) 9 WCT1/0/9 Clear (In use) 10 WCT1/0/10 Clear (In use) 11 WCT1/0/11 Clear (In use) 12 WCT1/0/12 Clear (In use) 13 WCT1/0/13 Clear (In use) 14 WCT1/0/14 Clear (In use) 15 WCT1/0/15 Clear (In use) 16 WCT1/0/16 HDLCFCS (In use) 17 WCT1/0/17 Clear (In use) 18 WCT1/0/18 Clear (In use) 19 WCT1/0/19 Clear (In use) 20 WCT1/0/20 Clear (In use) 21 WCT1/0/21 Clear (In use) 22 WCT1/0/22 Clear (In use) 23 WCT1/0/23 Clear (In use) 24 WCT1/0/24 Clear (In use) 25 WCT1/0/25 Clear (In use) 26 WCT1/0/26 Clear (In use) 27 WCT1/0/27 Clear (In use) 28 WCT1/0/28 Clear (In use) 29 WCT1/0/29 Clear (In use) 30 WCT1/0/30 Clear (In use) 31 WCT1/0/31 Clear (In use) [EMAIL PROTECTED] ~]# pri intense debug span 1, - attached as a text file due to its length. /var/log/asterisk/full - not found, only: [EMAIL PROTECTED] ~]# ls /var/log/asterisk/ cdr-csv cdr-custom event_log event_log.0 messages messages.0 queue_log queue_log.0 however the only usefull output of messages: [Apr 16 21:17:30] WARNING[9905] app_dial.c: Unable to create channel of type 'Zap' (cause 0 - Unknown) extensions.conf : [general] static=yes writeprotect=no clearglobalvars=no [globals] CONSOLE=Console/dsp ;
Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box,
there is no host entry and it uses dbsecret=dundi/secret, the
dundi.conf
priv = dundi-internal,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
[00:19:66:1C:78:D5] ; Dev Box
model = symmetric
host = 192.168.99.252
inkey = eus
outkey = eus
include = priv
permit = priv
qualify = yes
From iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=longdistance
Hope this helps, in your case Dundi will save you a world of work on
configuring that many systems, in fact if you structure Dundi like
spokes around a small number of master servers, the config gets real
easy.Let me know how it goes.
On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other box is
SIP/[EMAIL PROTECTED]
How can you use authentication, along with SIP, along with specifying
extension?
My sip.conf has a friend defined:
[priv]
host=dynamic
secret=priv
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal\
type=friend
I need to specify the sip channel to use the priv peer, priv secret, and
pass the extension. I've tried defining my mapping as:
Priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
But obviously the console on the far end complains that peer
a.b.c.d/${NUMBER} cannot be found.
Thanks for any insight into this. I'd prefer not having to define a sip
peer per box(I have 25 connected in my dundi cloud), nor would I like to
enable anonymous SIP calls, as I have the ports open to the world for
inbound sip from bandwidth.com
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Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Brian J. Murrell wrote: | | Yeah, well, that's all fine and dandy as long as more capacity is an | option. Many people are already subscribed to the most capacity | available to them and using it. | | b. Apparently man people don't understand that those QoS settings on routers mean little most of the time. Most providers resell QoS as a premium service, so while many waste their time painting their packets those markings get stripped. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQIVAwUBSAc5R4OeOV2sx4+mAQKEpQ//dYu+9MFaHgHzbBntTMbUHuY4usW5Aq+L crMlq3nYqgi8kWfVShhEozKHvtaYc7J7YBSkE2QprhM/YTp+wE3Oy9NM5GU6Ckhz IDaFNteO62zyxg5ljE81iIQd0tTJjutIf3FQVZBegzpINGIiEkjKBfbx/4UiO6HL bexoS3pnV4xjjS8xO8rMNl8+1XVubpG42K1/alw0G7y7W9Pog+u67+dLx1Tnx0EX RTlAeLZ64u5hy7CXeRdLSM3Onn8IuCnOIP2Py4OEUjLH8K4yMb83IVlhv+KSp4q4 5Tw7LWFsM/NZ0J6xz3MeUnXJHOkNK6Z5UJAfV1LmjiWdpxDCfYDifu6Y5D425+po gd/zHRI+SZJAhzN4l0oWIxSRQdCL6APyFqYFftO9bxAzDoK6EMXADIPvc3Ovb/A0 eUh6rZAe3y5/FfQy29GN23u5//ahFDCzQ9YqhbDjLEc/Z+PLi/lsEdWwWMrUMyus Q4nBs9osuxjRZYWEKUTLal+ItNL/BSiqHurN1T/l3W1/xigYiZHByxEBI2/+jYX6 66wQU6CSE2YC+n9R+rbsAP5OawOTxpXnDdTXEydHCPgdOAS5HmrwTp0t5MNZ4V/N iSGIBBAcV0HJIKRKaeGweIRGStAQPXbfQ9Qha7uYOqnyYwPbt18/vw08YlbdXXCO woCJ+I+AchI= =+Lgr -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] imap voicemail
5060? [EMAIL PROTECTED] ~]# netstat -an | grep 5060 udp0 0 0.0.0.0:50600.0.0.0:* how then do I tell it to use imaps? On Thu, Apr 17, 2008 at 2:38 PM, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: Hello. I'm trying to use gmail's imap feature w/ asterisk imap voicemail. I compiled c-client with the following settings: make lr5 IP6=4 and asterisk with: ./configure --with-imap=/usr/src/imap-2007a/ However if i enable any if the imap settings in voicemail.conf, asterisk starts acting funny and dosent allow any calls imapserver=imap.gmail.com imapport=993 mapfolder=Voicemail Where did I go wrong? Note that port 993 is IMAPS and I think you are telling it to use IMAP. Note also that Asterisk can get hung once in a while when using IMAP storage. Try netstat -an | grep 5060 and see whether you have a queue building there. __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Moshe Brevda, CTO ipconnect, ltd. 26 Strauss St., Jerusalem, Israel W. 1.800.800.456 (+9722.569.5295) M. +97254.666.1367 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
On Thu, 2008-04-17 at 07:54 -0400, sil wrote: Apparently man people don't understand that those QoS settings on routers mean little most of the time. Most providers resell QoS as a premium service, so while many waste their time painting their packets those markings get stripped. Maybe your understanding of QOS and mine is different. Of course I have no illusions that I can assign a priority to my packets that is going to be meaningful to anyone once they leave my network. But certainly at my choke point which is of course my Internet uplink, I can apply QOS (i.e. traffic shaping, which is what the OP's router was offering) to make sure that what little capacity is there is giving priority to my voice traffic. Think of my ISP uplink as that moderately congested road in which emergency vehicles need to have other casual traffic pull over and let it through. Traffic shaping is the effect of those vehicles pulling over and letting the voice traffic through in priority. This is exactly what OP's router was allowing him to do, albeit in what sounds like a really crappy way -- only 3 ports or something like that. b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
--- Nestor A. Diaz [EMAIL PROTECTED] wrote: Vieri wrote: Did you try a show channels to see if there were stale channels for peer 200? I had the same problem you describe but it was due to hung channels (used * 1.4.18.1 with rtp*timeout and saw inuse peers during the pre-timeout periods even though the agents weren't on a call). No, i don't , but how do do you fix this problem ? with rtp timeout ? rtp*timeout for sip peers is not a fix but a workaround. Try to set both values and reload sip. Then when you witness what you posted try doing a core show channels. You can then try to soft hangup a stuck channel or wait for the rtp*timeouts. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] buying cards from pakistan
Rizwan Hisham wrote: Hi all, i want to buy a pci or whatever card for asterisk to plug in my telephone line into it and use asterisk as a pbx. i have only one telephone line at home. can you recommend me a simple cheap card which i can buy in pakistan. I live in pakistan, and i dont know any dealers here who sell asterisk cards. if someone knows where to buy cards in pakistan, plz tell me about it. snip / I bought an X100p card from an eBay seller in the USA and I live in the UK. If you can use eBay or something similar why not do that? I am guessing that your PSTN will be similar to the rest of the world as most Telcos use one of only three vendors' switches. This one: http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItemitem=130213890766 looks O.K. albeit a little more expensive. This is the kind I bought as it has a low profile bracket so it would fit in a small PC case. If you use a card like this, get the OSLEC echo canceller and they work a treat! Hope this helps Alan (PS - There's some info on my blog - link below - about setting up a small asterisk home PC and getting OSLEC to work etc...) -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FSX gateways
Hi guys, What are some reliable sip to FSX gateways with four ports and eight ports? I've used some Linksys and Grandstream devices and I find that at unexplained times there will be echo on the line. Sometimes this happens on the end where the devices is placed and sometimes this happens on the other end. Also are there devices that support codecs such as ilbc or gsm so that I can put four to eight phones on a dsl line? Thanks for suggestions. Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Warning 2512
In article [EMAIL PROTECTED], Rizwan Hisham [EMAIL PROTECTED] wrote: -=-=-=-=-=- -=-=-=-=-=- I just saw the sip debug and its showing that for every notify request, asterisk is sending a bad request response. here is the debug --- SIP read from 70.80.000.00:1031 --- NOTIFY sip:69.90.111.11:9060 SIP/2.0 Via: SIP/2.0/UDP 70.80.000.00:1032;branch=z9hG4bK-ade549bd From: Blake sip:[EMAIL PROTECTED] [EMAIL PROTECTED] ;tag=90115683e082af23o0 The above two lines look very wrong to me. No wonder Asterisk is complaining! I'm not familiar with the Linksys SPA2102, but it looks like you have something wrong in its configuration of the SIP details. I think the From line ought to read something like this instead: From: Blake sip:[EMAIL PROTECTED];tag=90115683e082af23o0 Why it is trying to nest another [EMAIL PROTECTED] inside the SIP address, I don't know. Cheers Tony To: sip:69.90.111.11 Call-ID: [EMAIL PROTECTED] CSeq: 7741 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA2102-5.2.3 Content-Length: 0 - --- (10 headers 0 lines) --- --- Transmitting (no NAT) to 70.80.000.00:1031 --- SIP/2.0 489 Bad event Via: SIP/2.0/UDP 70.80.000.00:1032;branch=z9hG4bK-ade549bd;received= 70.80.000.00 From: Blake sip:[EMAIL PROTECTED] [EMAIL PROTECTED] ;tag=90115683e082af23o0 To: sip:69.90.111.11;tag=as3ef6a439 Call-ID: [EMAIL PROTECTED] CSeq: 7741 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Why is it doing so? On Thu, Apr 17, 2008 at 2:36 AM, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Rizwan Hisham [EMAIL PROTECTED] wrote: I have been seeing a lot of the following warning messages on my asterisk cli. Can naybody tell why these messages are showing up. I am using only SIP to make calls from m asterisk. [Apr 17 04:52:24] WARNING[2512]: chan_sip.c:6480 determine_firstline_parts: Bad request protocol Bad event Also it will be great if anybody can tell where i can find the explanation of all the warnig codes and error codes of asterisk if there is any. The [2512] is not a warning code. It is just the process ID of the Asterisk process or thread that generated the warning. The next part of your message (chan_sip.c:6480) shows the source file and line number where the error was generated. You can go to that point in the file to see what kind of checks it was making. You can also turn on SIP debugging at the Asterisk CLI prompt to see the packets sent to/from Asterisk. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Rizwan Hisham -=-=-=-=-=- [Alternative: text/html] -=-=-=-=-=- -=-=-=-=-=- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=- -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Brian J. Murrell wrote: | Maybe your understanding of QOS and mine is different. Of course I have | no illusions that I can assign a priority to my packets that is going to | be meaningful to anyone once they leave my network. | | But certainly at my choke point which is of course my Internet uplink, I | can apply QOS (i.e. traffic shaping, which is what the OP's router was | offering) to make sure that what little capacity is there is giving | priority to my voice traffic. | | Think of my ISP uplink as that moderately congested road in which | emergency vehicles need to have other casual traffic pull over and let | it through. Traffic shaping is the effect of those vehicles pulling | over and letting the voice traffic through in priority. This is exactly | what OP's router was allowing him to do, albeit in what sounds like a | really crappy way -- only 3 ports or something like that. | | b. Let's take a bare bones look at this. Let's say your connection is 300k and you have five packets coming in at 60k each to saturate your network: Provider to you Packet 1 You Packet 2 You Packet 3 You Packet 4 You Packet 5 You You believe that this is happening: Packet 1 You --- This is voice send it first -- Device Packet 2 You --- This is voice send it first -- Device Packet 3 You --- This is P2P leave it 4 last -- Device Packet 4 You --- This is P2P leave it 4 last -- Device Packet 5 You --- This is AIM make it second! -- Device Its fine and dandy, but the problem is you're still getting 5 packets. You're still saturated period. No QoS in the world outside of your provider and more bandwidth can alleviate that. Your provider is not going to care what you do once its passed to the CPE. So look at it logically again. QoS on a home router... Useless COMING IN. Going out... Means little but helps MINIMALLY. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQIVAwUBSAdCdoOeOV2sx4+mAQImCA//Q3pXHy/hUUqc/RvN/WNXzYiXMVR5gmKR xdNc+GZkN0ks16wKqJLxXITDwXE9vWygEmY3G97xo9f3jFR/NihtiTDTo7n/nvA6 GDC1gOw5UY20793ACHdL4mroCL7A8UMUdZGDZyyhQVSIpKZ4Uhk9bwgDPzRCYkOp rwgL2WyQAPrk6GjKg/XIg3H6vBtI6ZcuXV5xu5CoPxOb1hPEzj/AX/OiQbZlAGPY CLFWnVSs1YBM4rq2Jt3KA7kKPsFST81JMMWSxU+axKzmaa6LmU29FgX4WG8jBG5s 0Nxk0PkXIzu6XfLVkU8Dop5FCUpxbDRmh6OyXyvluQ2SEBh48ZiPSnClDI+Ue9JN J5z2QQen8qtK/HdbCDp08MF6MSiEceYYCcwWHGMg9KlD3u2FgY9rrPZ3hKrP9Tz5 1ciLXig5mvyhWBGIS5mIhg7QnnWAzMsXjbQ8buHgir82ptDbM3wSyWdkWHNR47Fr uFe+QGVV4JHFzHsDkeo/qfGA2juwazMfNXJyV67vWnyZNhnhtZ+kEAbMXeABvhjQ rw/bgtq7gdiv/fwIgq51WKEPQbyHozRpqdyZPUzBJBsDND5iKivzIboug+hS4QJH tqK/c29mir/0D5CbXswhCTjbiUIYIyH8Gu+OU3G1uhNrv+TRm7E+8jCtvM+zfvXu AaTY8D7BmJo= =ldzm -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi and SIP
I have it working via IAX, when I try changing everything to SIP I can't
specify a username and an extension, so it becomes useless.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Thursday, April 17, 2008 6:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi and SIP
Jeremy,
Here is a working sample to compare to. This is an IAX2 setup, but the
only difference is in the mapping change IAX2 to SIP. Notice the 4th
setting in the mapping? It defines to use the IAX2 peer priv with
the secret generated of the key defined in the peers section of
dundi.conf. When you look at the peer in iax.conf on the remote box,
there is no host entry and it uses dbsecret=dundi/secret, the
dundi.conf
priv = dundi-internal,0,IAX2,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
[00:19:66:1C:78:D5] ; Dev Box
model = symmetric
host = 192.168.99.252
inkey = eus
outkey = eus
include = priv
permit = priv
qualify = yes
From iax.conf
[priv]
type=friend
dbsecret=dundi/secret
context=longdistance
Hope this helps, in your case Dundi will save you a world of work on
configuring that many systems, in fact if you structure Dundi like
spokes around a small number of master servers, the config gets real
easy.Let me know how it goes.
On Wed, Apr 16, 2008 at 8:41 AM, Jeremy Mann [EMAIL PROTECTED] wrote:
I'm a little confused with DUNDi and SIP as the backend channel type:
Dundi.conf:
[mappings]
priv = dundi-priv-local,0,SIP,[EMAIL PROTECTED],nopartial
Using the above, the dial string passed to the person on the other box is
SIP/[EMAIL PROTECTED]
How can you use authentication, along with SIP, along with specifying
extension?
My sip.conf has a friend defined:
[priv]
host=dynamic
secret=priv
disallow=all
allow=ulaw
canreinvite=no
nat=no
context=from-internal\
type=friend
I need to specify the sip channel to use the priv peer, priv secret, and
pass the extension. I've tried defining my mapping as:
Priv = dundi-priv-local,0,SIP,priv:[EMAIL PROTECTED]/${NUMBER},nopartial
But obviously the console on the far end complains that peer
a.b.c.d/${NUMBER} cannot be found.
Thanks for any insight into this. I'd prefer not having to define a sip
peer per box(I have 25 connected in my dundi cloud), nor would I like to
enable anonymous SIP calls, as I have the ports open to the world for
inbound sip from bandwidth.com
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Bruce Reeves, dCAp
EUS Networks
Office: 212-624-5943
Web: www.euscorp.com
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This e-mail, facsimile, or letter and any files or attachments transmitted with
it contains information that is confidential and privileged. This information
is intended only for the use of the individual(s) and entity(ies) to whom it is
addressed. If you are the intended recipient, further disclosures are
prohibited without proper authorization. If you are not the intended recipient,
any disclosure, copying, printing, or use of this information is strictly
prohibited and possibly a violation of federal or state law and regulations. If
you have received this information in error, please notify Texas Health
Management Group immediately at 1-817-310-4999. Texas Health Management Group,
its subsidiaries, and affiliates hereby claim all applicable privileges related
to this information.
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Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
On Thu, 2008-04-17 at 08:36 -0400, J. Oquendo wrote: Brian J. Murrell wrote: | But certainly at my choke point which is of course my Internet uplink, ^^^ I | can apply QOS (i.e. traffic shaping, which is what the OP's router was | offering) to make sure that what little capacity is there is giving | priority to my voice traffic. Let's take a bare bones look at this. Let's say your connection is 300k Downstream or upstream? Notice I said Internet uplink in my previous message. Anyone at all familiar with traffic shaping understands that they can only shape the uplink, not the downlink. The best you can do with the downlink is to police it to try to keep the congestion below 100%. But that's mostly alright given how the ISPs have perverted the Internet with asymmetric last mile connections to consumers. and you have five packets coming in at 60k each to saturate your network: First of all,your whole example is pointless as you are clearly talking about downstream and I have already said that anyone knowledgeable with traffic shaping knows you cannot shape the downlink only the uplink. However, let's see where else your example fails. My MTU is only about 1500 bytes or so, so 60k packets to me are impossible. I'd tend to guess that for most of the Internet, packets max out at about 1500 given the prevalence of ethernet connected devices. So in order to saturate my 300k you'd have to send me 200 packets all in that one second. Provider to you Packet 1 You Packet 2 You Packet 3 You Packet 4 You Packet 5 You You believe that this is happening: Packet 1 You --- This is voice send it first -- Device Packet 2 You --- This is voice send it first -- Device Packet 3 You --- This is P2P leave it 4 last -- Device Packet 4 You --- This is P2P leave it 4 last -- Device Packet 5 You --- This is AIM make it second! -- Device As I've said, you cannot shape this traffic. I've already conceded that. But again, OP was talking about uplink shaping, not downlink. Its fine and dandy, but the problem is you're still getting 5 packets. You're still saturated period. Right. You cannot shape the downlink. You can only police it to prevent packet loss. No QoS in the world outside of your provider and more bandwidth can alleviate that. If more bandwidth is an option, but I already stated that for many people, it's not an option. They have exactly one or two choices and they are subscribed to their maximum available. QoS on a home router... Useless COMING IN. Going out... Means little but helps MINIMALLY. Not at all little. If you have a lot of low priority outgoing traffic (i.e. p2p) saturating your link, uplink traffic shaping will mean the difference between a completely unintelligible call and something very acceptable. b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Wed, Apr 16, 2008 at 7:18 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: Ex Vito wrote: Tested with no 4K stack kernel and stackcleanup svn branch zaptel version. Correct, the kernel no longer complains about the soft hangup. However the system still hangs (console inoperative, etc) while ztcfg'ing... That is normal while the firmware is loading. It should go away after the firmware has loaded. Ok. So here is our reasoning according to collected info. Please correct us where appropriate: 1. The system is supposed to hang while the firmware loads into the DSPs under any zaptel version 2. zaptel 1.4.10 leads to a soft hangup detected, zaptel 1.4.9.2 does not (assuming softhangup detection active in kernel) 3. zaptel 1.4.10 takes much longer ztcfg'ing than 1.4.9.2, that's why the soft hangup is detected under zaptel 1.4.10 (difficult to time, but let's say 1.4.10 takes 10s, 1.4.9.2 takes 3s) Now, back to the original question: - Should this be considered a regression ? - Next steps: a) file a bug and move this analysis to the bug tracker b) don't file bug and move analysis to the dev list c) don't file bug, keep on working on the users list I recommend 1.4.10 by default. However, from what you said it would appear that you are having problems with 1.4.10 so you might stay with 1.4.10 if you are not having any issues with it. Did you mean this instead ? ...so you might stay with 1.4.9.2 if you are not having any issues with it. We think so. Again, thanks for clarifying. Cheers, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Click-to-call client
I think videoreps.net It´s not free. But, I discover that I really need is click-to-talk, excuse me. On Wed, Apr 16, 2008 at 5:05 PM, Bob G [EMAIL PROTECTED] wrote: Introducing Click-to-Call http://1ezphone.com/ Posted: 16 Apr 2008 9:55 AM PDT The 1EZphone browser softphone has created so much buzz in the media that a lot of individual users and companies who have a web-presence; Websites, Online Advertising, Blogs, Customer support etc have asked for a Click-to-Call service. The 1Ezphone web-based Click-to-Call service is based on our browser VoIP lite technology that allows users to make and receive phone calls from any browser without the need to download software. The Click-to Call API can be embedded on any Website, E-mail, and Online Advertisement when a user clicks your object they immediately call your salesperson or customer service representative telephone number and speak to your agent over their PC. Building a reliable Click-to-Call requires substantial amount of knowledge in VoIP, and a good backend infrastructure. The good news is that now it is easy add Click-to Call to any online service in just a few minutes with just a few lines of code using 1ezphone's. Since the release of our APIs, we got several requests from companies and developers who were interested in knowing in building their own Click-to-Call service directly to their SIP servers. You can have the button/widget running through the 1Ezphones servers without getting into the complex world of VoIP or any expensive setup or build a service to your own backend infrastructure. If you are interested in adding Click-to-Call for your customers or building your own Click-to Call system please contact 1ezphone at [EMAIL PROTECTED] - Original Message - From: BJ Weschke To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best Click-to-call client Date: Wed, 16 Apr 2008 11:37:40 -0400 equis software wrote: Hi, I need to make Click-to-Call web application to connect with an asterisk server. I´m using Java What solution recommend me? I did a spiel on this at Astricon last year. The slide deck is somewhere around for those interested, but now we also have some code to show for it. :-) Take a look at this developer branch at http://www.asterisk.org/node/48440 and then we've put some pieces together for the Java side of things using Ignite's Realtime API for messaging. http://svn.btwtech.com/svnview/coolvocals/trunk/cti-server/click2call/ http://svn.btwtech.com/svnview/coolvocals/trunk/cti-client/click2call/ Basically the idea here is that there's a servlet that honors requests into it (think AJAX Remote calls from the browser) and then turns around and puts that request into a jabber message that goes to a centralized Servlet that can proxy requests across multiple servers (scalability/LCR/etc) and that in turn launches an Originate call in to the AMI of the machine that was decided would receive the request. Once that hand off is done, the proxy machine that received and directed the original request is now out of the middle of things and jabber messages are sent directly back to the client to signal call progress of the click to call. Is it a shrink wrapped and ready to go package that's completely documented and involves no technical knowledge whatsoever for implementation? U.. no, but that might happen in the relatively near future. :-) What it IS though is solid working code (yes, it has been fully unit tested out and is functional) contributed back to the community so we can all start to make something with it if we so choose. If there's enough interest, I'd certainly entertain opening up a blog site and open up the branch of the Java code for community contributions as well in addition to doing a more detailed tutorial on usage of the code at the upcoming Astricon this year. BJ -- -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Want an e-mail address like mine? Get a *free e-mail *account today at www.mail.comhttp://www.mail.com/Product.aspx ! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Brian J. Murrell wrote: | Not at all little. If you have a lot of low priority outgoing traffic | (i.e. p2p) saturating your link, uplink traffic shaping will mean the | difference between a completely unintelligible call and something very | acceptable. Is it? So you're telling me if you're saturated on the way in, fixing up your packets on the way out is the solution. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQIVAwUBSAdPUoOeOV2sx4+mAQJT6w/7Bm4hcyAaaLlwlYo8Dsfw5oUCUOW+0TLy jrIWS1piOYbe+MBjpoliOF7nJETrQnFUP5y/kZjjOTxCuyz1XLlj7ExOdddGXudp My2tO81/Gkx/kicOQDtdxLKtMBWI8ix3Ef1Z1dtQIh4DYJqSGbzgTez5D1WfxhXG IZ2IFq8CKzpjT7oAExmo/l7QqetCXPMgM4gZ24CDXlESL/esYBJL5sWfzG+4dLyJ 4INMpPnckXjdf/WyCIeMDrGRAEKpNQ8Ls+X/EAgwqJ83Z6iTJUrW6xMfO9KXAlDP BNwrX1/Xlx0quNd+tH+u0j8DcQ0sy9jt4KixOQYqCb9VtpDz5Ucf8zyqMC277ugz FwaDSpUlkASe/JK0m/IFf4lvnrgBna1jDFa5k13u8R+Ja1rcb0+S7I5Rk6MxpBCo xIfRIGHqO/hmAv3ckj2qIoGetlPZNTT94fgGV/d5UnAU4eOTuNXeZURS9Wf3XVN6 Yc90oGHKWfB3O0XJNS/QI4LeI7BxWJUDmyC1PczKfhIj9ox9K+GD1tSvto3nSqZE NFpdcG7Ch1EDAYZuptvAp+3tKy+ifLYmultAq7/ehBeJ+t0GJxxwFqIYTGwuUCBh M0Gd570V4baOhl3UY917uwkTb4bBXS+9wh2J7qTUqVmGlOYC/6x0MrJLabetQiKT A5+VziQWa/Y= =tSIZ -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
On Thu, 2008-04-17 at 09:25 -0400, J. Oquendo wrote: Is it? So you're telling me if you're saturated on the way in, fixing up your packets on the way out is the solution. I think I've made it clear that my argument is only about uplink shaping and the requirement for it given the asymmetric nature of a lot of last mile connections existing today. Funny enough that is *exactly* what the OP was asking about. b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chanspy on Asterisk 1.4.19
Steve Rawlings wrote: exten = 596,n,ChanSpy(|g(2000)) ...snip... This worked fine with 1.4.18.1. With 1.4.19 if I dial 596 I get answered but there's no spying, the only way I could get this to work was with - exten = 596,n,ChanSpy(|b) but this spied on all channels, not just those with SPYGROUP set to 2000 so not much use to us. You can pass multiple options to a dialplan application, so instead of downgrading ChanSpy, you could have just done: exten = 596,n,ChanSpy(|bg(2000)) Or am I missing something? -- Sean Bright [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Brian J. Murrell wrote: | I think I've made it clear that my argument is only about uplink shaping | and the requirement for it given the asymmetric nature of a lot of last | mile connections existing today. Funny enough that is *exactly* what | the OP was asking about. | | b. Answers the question with minimal relevance, not even a band-aid solution. You fixing up inbound traffic will do nothing for a horrible conversation if you're congested coming in. Solution would be to add more bandwidth. Else you could fiddle around around creating all the fuzzy rules on the planet shaping traffic all sorts of methods once its in your CPE but this WILL NOT HELP YOU HAVE A BETTER CONVERSATION. When it does, when someone can realistically point this out please let me know so I can switch from a DS3 to T1 and save money. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQIVAwUBSAdXC4OeOV2sx4+mAQIVFw/+NttSjlj132/ikQZN4pI6kJJH49GxJiMp aA9ugBu0jA9ZXSgU8oHw9ZbgkZfjalM1vtmekOW+w4eXUwlx82jEGEJ1e7iBT30e wB9cOOMlpn1+sFrZxesAz/8a7ziFC02Ydf2+V3j8FfPga8DuHWtF/hubm7p/d3Zy Km1Vm1ruajCTM9PAvVO/Jj2TybzYwWj7Pj2TzwZEsYCXUuj5/E0fnQjJKCI4q6e6 UHsOs5tpqedzRCSJ2Zv96xkHAFWDLOUke2vXp20ZETnOxqOVtULm+EuYXsHvauYN 6sMZf7Tq04+jMrbR1GWLCevvEoJN1XpTEOBb3yv7S/7U7Ih/mQfluHNj4hVUACYs vFlIJyHBeLxeAOH5VFm66SDtIQ2TKGLuFblDD5E6MmhXYdhwdwsmGecfaEJHR/+K 83CDQ1P1tDtN6JjcYXsoN8125uRKYH2EQunfZq01GJQlj6QNJcZHcv9FrRYXan42 7yxB+h1UpgNLMAthOQsQ8+nt7rRD8v0GlPZBwXlRF1n2S2jAVJiwlrihfiW5xA6C LsRuU7GIo/XkX/zNQk2BIGszziIEGcYaJjYnXBdsP2QN6IwkCz8xwQbgtssSMqmd kbFelVI4BepzbG2lUVkmUAavoFL7T1c9eIyMU9vunOJtP/azTadXP9ITS936mKYK pYOvun1cnqU= =I0yU -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Thu, Apr 17, 2008 at 2:36 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Apr 17, 2008 at 02:20:57PM +0100, Ex Vito wrote: - Should this be considered a regression ? Yes, it is a regression, and thus a bug. Mattf has already offered you to work with him on resolving this. FYI, Submitted bug 0012468 as per Tzafrir suggestion. (http://bugs.digium.com/view.php?id=12468) -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
May I suggest the following read: A Beginners Guide To Successful VOIP Over DSL http://www.smallnetbuilder.com/content/view/30340/83/ Which covers both QoS and traffic shaping in small routers. It was written based upon my own experience with both Asterisk and hosted PBX providers. Michael On Thu, 17 Apr 2008 07:16:40 -0400, sil wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Simon wrote: | Is this worth doing? If so, what ports should i specifiy? http://www.bricklin.com/qos.htm -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQIVAwUBSAcxA4OeOV2sx4+mAQKrKg//Vs5n0K1m1+C+sTSol2a1MbFHU/QCh1fT u5IwLOvQwcDvxOHikYYk8Ornm5FEbp4cewnCVKS9BeLkurlaQ1qXiwPdCHhnrNhn q2ufIskPudXn2cNj9pbAkZhmNL7R7m1XruITBGrXvV4d2WAZy6lmNGnVOhipU5ff HIV1aAacawCJG6oJjpD/NvjydHYwP6KgvOJkcLICrb7FEC4bfDfULQxThFF9Jzf3 aMzvddM+GdBGzhT0q7FH7JGQmuahlN1kdIyLY8Rw+/ouEgm4xYeyZ486JaBk2xOK 5ZnYqXXLmNuB7LPIkSCp2Fi5usFUBrKq8nanbvonw9Te3pILCG/FAfg/+O3Y2ZQb 0aZsUW13BHj9hfiZYLnKGCeJV1hLLuLWH+fP7E9kzFbi8ls7/Ke+oe7l8fgRFfzt oaInPjl1tshzbaOesSX1H8OI5QyfGgmuhyVu5E0tFmy9HX7QnxxBrI/GXMwQ3F6/ +Qv058sJ5qjQtGMi0fI6GoDa3xQRCzyBgWjuOBHhk64FVnMs3Rdoti69YhsWH52a LQMleyChhVQ0nrP5eVaykEryLNRw7jDV1X/ivtPNOHfQ0fN8337AHPmmKpzo22sK xdzK+RY1I/qZ+SOD6YPaiKjxaB9gqDPe7jGy41NlGsjnrgUjJh2c2/tvcyDYaW4u 0J5kiHsMXLI= =oY+k -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip or IAX device with professional balanced audio out
Hi all, I've been googling for a solution here and haven't really come up with anything yet. We're doing an Asterisk install for a local radio station, and we're looking for a phone that they can use in their control room hooked up to their mixer board for recording calls. So, when you phone in for some contest or to request a song they record it and play it back a few minutes later on the air. They are currently recording calls from a hacked pots phone, but I was hoping for something a little more elegant with their new system. Has anyone run across a solution that might work nice here, or is there some other way of tackling this problem that I may have overlooked? Thank for your suggestions. Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Differents routes for differents extensions
Hi everybody, I need to use different outbound routes from calls started by different extensions; I mean, that the extension A when dialing 011543... has to get access always on the 1st trunk, the extension B when dialing another number has always to access the outside world on the 2nd trunk, and so on. Some kind of solution I thought involved the use of a fake dialcode, whic is prepended to the dial number and then stripped from an Outbound route section (and then the trunk is dialed): ext. A call 011543... prepend 41 --- Outbound route for 41|. --- Appropriate trunk dial . The only matter is that I have NO clue on where to append this code for outgoing calls from these specific extensions. If anyone has a simpler idea (yeah, mine is pretty twisted %) ) or knows how to put a code before the dial string of an extension, let me know! Thanks in advance, Marco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip or IAX device with professional balanced audio out
How about avoiding the phone entirely in the playback phase? Have asterisk record the call to disk in MP3 or Slin, then use a pc with decent audio card to read it off the shared disk and feed it to the mixer. Tim. - Original Message - From: Bob Pierce [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 17, 2008 3:45:56 PM (GMT) Europe/London Subject: [asterisk-users] Sip or IAX device with professional balanced audio out Hi all, I've been googling for a solution here and haven't really come up with anything yet. We're doing an Asterisk install for a local radio station, and we're looking for a phone that they can use in their control room hooked up to their mixer board for recording calls. So, when you phone in for some contest or to request a song they record it and play it back a few minutes later on the air. They are currently recording calls from a hacked pots phone, but I was hoping for something a little more elegant with their new system. Has anyone run across a solution that might work nice here, or is there some other way of tackling this problem that I may have overlooked? Thank for your suggestions. Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chanspy on Asterisk 1.4.19
On Thu, Apr 17, 2008 at 9:44 AM, Sean Bright [EMAIL PROTECTED] wrote: Steve Rawlings wrote: exten = 596,n,ChanSpy(|g(2000)) ...snip... This worked fine with 1.4.18.1. With 1.4.19 if I dial 596 I get answered but there's no spying, the only way I could get this to work was with - exten = 596,n,ChanSpy(|b) but this spied on all channels, not just those with SPYGROUP set to 2000 so not much use to us. You can pass multiple options to a dialplan application, so instead of downgrading ChanSpy, you could have just done: exten = 596,n,ChanSpy(|bg(2000)) Or am I missing something? -- Sean Bright [EMAIL PROTECTED] Should one have to change their dialplan for functionality to remain the same in the same version? I thought it was only really supposed to change when something is deprecated (and documented in a README or something after throwing warnings for a version or so.) Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Differents routes for differents extensions
Create different contexts and assign them to the extensions [trunk1] exten = .X,1,Dial() [trunk2] exten = .X,Dial() and in sip.conf or iax.conf [exten1] ... context = trunk1 [exten2] context=trunk2 Marco escribió: Hi everybody, I need to use different outbound routes from calls started by different extensions; I mean, that the extension A when dialing 011543... has to get access always on the 1st trunk, the extension B when dialing another number has always to access the outside world on the 2nd trunk, and so on. Some kind of solution I thought involved the use of a fake dialcode, whic is prepended to the dial number and then stripped from an Outbound route section (and then the trunk is dialed): ext. A call 011543... prepend 41 --- Outbound route for 41|. --- Appropriate trunk dial . The only matter is that I have NO clue on where to append this code for outgoing calls from these specific extensions. If anyone has a simpler idea (yeah, mine is pretty twisted %) ) or knows how to put a code before the dial string of an extension, let me know! Thanks in advance, Marco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME Cryptographic Signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sometimes UNREACHABLE!/REACHABLE doesn't appear in the log when it should
Hi, everyone. I'm having a problem with qualify=yes sip.conf option. Sometimes, when a device registered with asterisk goes offline, I'm not getting a message about it in /var/log/asterisk/messages log. Sometimes the same happens with REACHABLE message, when a device comes back online. I'm pretty sure asterisk is aware of a device's current state. I can see it with sip show peers command. Is this a bug? If not, how could I achieve the desired behavior: every time a device changes its state, a message in the log appears about it. Thanks in advance and sorry for my English. I'm still learning :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
My personnal experience is if you`re looking for an inexpensive solution (SOHO), StreamEngine based routers (a lot of D-Link products are Streamengine based, for example the DI-724GU and the DIR-655) do a decent job of providing QoS on the upstream, which is where you (usually) need it anyways. And the good thing is you often do not have to do anything but set the upload bandwidth (yes there is an automatic mode, but it's not that great). Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Thursday, April 17, 2008 10:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway! May I suggest the following read: A Beginners Guide To Successful VOIP Over DSL http://www.smallnetbuilder.com/content/view/30340/83/ Which covers both QoS and traffic shaping in small routers. It was written based upon my own experience with both Asterisk and hosted PBX providers. Michael On Thu, 17 Apr 2008 07:16:40 -0400, sil wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Simon wrote: | Is this worth doing? If so, what ports should i specifiy? http://www.bricklin.com/qos.htm -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQIVAwUBSAcxA4OeOV2sx4+mAQKrKg//Vs5n0K1m1+C+sTSol2a1MbFHU/QCh1fT u5IwLOvQwcDvxOHikYYk8Ornm5FEbp4cewnCVKS9BeLkurlaQ1qXiwPdCHhnrNhn q2ufIskPudXn2cNj9pbAkZhmNL7R7m1XruITBGrXvV4d2WAZy6lmNGnVOhipU5ff HIV1aAacawCJG6oJjpD/NvjydHYwP6KgvOJkcLICrb7FEC4bfDfULQxThFF9Jzf3 aMzvddM+GdBGzhT0q7FH7JGQmuahlN1kdIyLY8Rw+/ouEgm4xYeyZ486JaBk2xOK 5ZnYqXXLmNuB7LPIkSCp2Fi5usFUBrKq8nanbvonw9Te3pILCG/FAfg/+O3Y2ZQb 0aZsUW13BHj9hfiZYLnKGCeJV1hLLuLWH+fP7E9kzFbi8ls7/Ke+oe7l8fgRFfzt oaInPjl1tshzbaOesSX1H8OI5QyfGgmuhyVu5E0tFmy9HX7QnxxBrI/GXMwQ3F6/ +Qv058sJ5qjQtGMi0fI6GoDa3xQRCzyBgWjuOBHhk64FVnMs3Rdoti69YhsWH52a LQMleyChhVQ0nrP5eVaykEryLNRw7jDV1X/ivtPNOHfQ0fN8337AHPmmKpzo22sK xdzK+RY1I/qZ+SOD6YPaiKjxaB9gqDPe7jGy41NlGsjnrgUjJh2c2/tvcyDYaW4u 0J5kiHsMXLI= =oY+k -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sometimes UNREACHABLE!/REACHABLE doesn't appear in the log when it should
The asterisk code is full of fun things where it checks for things like that in multiple places but doesn't always handle every instance of the same check in the same way. This is getting resolved piecemeal and will eventually be minimized as the application develops, but I do not think things like this will ever completely go away. fadey wrote: Hi, everyone. I'm having a problem with qualify=yes sip.conf option. Sometimes, when a device registered with asterisk goes offline, I'm not getting a message about it in /var/log/asterisk/messages log. Sometimes the same happens with REACHABLE message, when a device comes back online. I'm pretty sure asterisk is aware of a device's current state. I can see it with sip show peers command. Is this a bug? If not, how could I achieve the desired behavior: every time a device changes its state, a message in the log appears about it. Thanks in advance and sorry for my English. I'm still learning :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
Mike wrote: do a decent job of providing QoS on the upstream, which is where you (usually) need it anyways. QOS can only be on outgoing, you can't set the priority of a packet after you receive it. The only other solution would be the cooperation of the ISP to provide QOS upstream of you. Good luck. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
ok, thanks, does rtp*timeout work if i have canreinvite=yes ? since rtp traffic is not passing thought asterisk, or i have to put canreinvite=no ? slds. rtp*timeout for sip peers is not a fix but a workaround. Try to set both values and reload sip. Then when you witness what you posted try doing a core show channels. You can then try to soft hangup a stuck channel or wait for the rtp*timeouts. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
On Thu, 2008-04-17 at 11:40 -0400, Chris Mason (Lists) wrote: Mike wrote: do a decent job of providing QoS on the upstream, which is where you (usually) need it anyways. QOS can only be on outgoing, Which is what he meant when he said upstream I believe. you can't set the priority of a packet after you receive it. Indeed. The only other solution would be the cooperation of the ISP to provide QOS upstream of you. Good luck. Heh. Yeah, no doubt. b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] status line header
Hello,
I need to read the Status-Line (I need to know if it's 603, 503, 404)
after a Dial. I have tried:
exten = s,2,Dial(SIP/[EMAIL PROTECTED],,tTwW)
exten = s,3,Set(t=${SIP_HEADER(Status-Line)})
But t is empty
I have also tried:
exten = s,5,Verbose(*** STATUS: ${DIALSTATUS})
exten = s,6,Verbose(*** STATUS2: ${HANGUPCAUSE})
DIALSTATUS: I usually get Congestion (but I know from sniffer that
sometime sStatus-Line is 404, sometmies 603 and sometimes 503).
HANGUPCAUSE: I have used in ISDN but here is always 0.
How could I get the Status-Line code?
Thank you very much,
--
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http://pinux.info Manresa - Barcelona
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Re: [asterisk-users] Differents routes for differents extensions
That makes PERFECT sense and also makes me aware that I need to review asterisk theory :-P I'll put it under test and let you know how it works. Thanks a lot! Marco Rodrigo Gonzalez ha scritto: Create different contexts and assign them to the extensions [trunk1] exten = .X,1,Dial() [trunk2] exten = .X,Dial() and in sip.conf or iax.conf [exten1] ... context = trunk1 [exten2] context=trunk2 Marco escribió: Hi everybody, I need to use different outbound routes from calls started by different extensions; I mean, that the extension A when dialing 011543... has to get access always on the 1st trunk, the extension B when dialing another number has always to access the outside world on the 2nd trunk, and so on. Some kind of solution I thought involved the use of a fake dialcode, whic is prepended to the dial number and then stripped from an Outbound route section (and then the trunk is dialed): ext. A call 011543... prepend 41 --- Outbound route for 41|. --- Appropriate trunk dial . The only matter is that I have NO clue on where to append this code for outgoing calls from these specific extensions. If anyone has a simpler idea (yeah, mine is pretty twisted %) ) or knows how to put a code before the dial string of an extension, let me know! Thanks in advance, Marco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] questions running 2 asterisk under the same LAN
Hi, I want to know if I am running two machines each with its own Asterisk on my LAN, show I change the port of one of the Asterisk to something like 5061? Otherwise, how does an external SIP client ( like IPkall.com) knows how to route DID call to Asterisk? What is the solution for this kind of setup? Thank you very much in advance for your inputs. Regards, Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 license count...
I need a refresher course on how many licenses I need to buy. I have an Asterisk server that receives calls by SIP (G729) and then sends them to the PSTN via 32 Zap interfaces on an Astribank. I cannot remember if the license is per channel or per call so I do not know if I need 32 or 64 licenses for this application. Could anyone please remind me? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chanspy on Asterisk 1.4.19
Steve Totaro wrote:
Should one have to change their dialplan for functionality to remain
the same in the same version?
I wasn't suggesting it wasn't a regression, just making the OP aware
that he can pass multiple arguments to a dialplan application (i.e.
ChanSpy(|bg(2000)))
He mentioned that he was able to get it to work in 1.4.19 by passing the
bridge argument ('b') but didn't seem to be aware that he could also
pass his original argument list ('g(2000)') as well. Seems easier to
just work around the problem with the additional argument than to
backport the application.
--
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[EMAIL PROTECTED]
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Re: [asterisk-users] G729 license count...
Afaik its per encode / decoder pair. In this case you will need 32 simultaneous encoders / decoders between g729 and slin, so you would need 32 licenses. Contact digium sales/support directly and you will know for sure :) Zoa Carlos Chavez wrote: I need a refresher course on how many licenses I need to buy. I have an Asterisk server that receives calls by SIP (G729) and then sends them to the PSTN via 32 Zap interfaces on an Astribank. I cannot remember if the license is per channel or per call so I do not know if I need 32 or 64 licenses for this application. Could anyone please remind me? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 license count...
http://store.digium.com/productview.php?product_code=G729CODEC http://www.digium.com/en/docs/G729/g729policy.php http://www.voip-info.org/wiki-Asterisk+G.729+Licensing On Thu, Apr 17, 2008 at 11:14 AM, Carlos Chavez [EMAIL PROTECTED] wrote: I need a refresher course on how many licenses I need to buy. I have an Asterisk server that receives calls by SIP (G729) and then sends them to the PSTN via 32 Zap interfaces on an Astribank. I cannot remember if the license is per channel or per call so I do not know if I need 32 or 64 licenses for this application. Could anyone please remind me? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] questions running 2 asterisk under the same LAN
On Thu, Apr 17, 2008 at 12:03 PM, Pete Kay [EMAIL PROTECTED] wrote: Hi, I want to know if I am running two machines each with its own Asterisk on my LAN, show I change the port of one of the Asterisk to something like 5061? Otherwise, how does an external SIP client ( like IPkall.com) knows how to route DID call to Asterisk? What is the solution for this kind of setup? Thank you very much in advance for your inputs. Regards, Pete Maybe if you draw a clearer picture of what you have/need. I guess you only have one public IP and everything is behind a NAT router? If you have or can get additional IPs, it would make port forwarding rules on your router easier (if it supports multiple IPs) than changing ports on the Asterisk boxen, that would probably be my approach if possible, from what I can gather from your description. Thanks, Steve Totaro This link might be of help. http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] users.conf and voicemail
Is there a way to specify per user attachment options for voicemail, from within users.conf? I know I can enable or disable it globally in voicemail.conf, but I have certain users that like the attachment feature, and others that don't. Also, can you enable/disable per user the deletion if it's attached? Thanks. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] questions running 2 asterisk under the same LAN
you surely are using port forwarding right now, something like this: wan(5061) -you_router_here-- lan(5061)---Asterisk_box_1(5061) so you only need to add this : wan(5062) -you_router_here-- lan(5061)---Asterisk_box_2(5061) just tell your provider that second account goes to port 5062. another solution would be to add a second connection to internet use it for second box as internet is cheap these days. On Thu, Apr 17, 2008 at 4:03 PM, Pete Kay [EMAIL PROTECTED] wrote: Hi, I want to know if I am running two machines each with its own Asterisk on my LAN, show I change the port of one of the Asterisk to something like 5061? Otherwise, how does an external SIP client ( like IPkall.com) knows how to route DID call to Asterisk? What is the solution for this kind of setup? Thank you very much in advance for your inputs. Regards, Pete ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] questions running 2 asterisk under the same LAN
On Fri, 18 Apr 2008, Pete Kay wrote: I want to know if I am running two machines each with its own Asterisk on my LAN, show I change the port of one of the Asterisk to something like 5061? It is common to have multiple instances of Asterisk listening to the same port number on the same LAN. Otherwise, how does an external SIP client ( like IPkall.com) knows how to route DID call to Asterisk? What is the solution for this kind of setup? It depends on the provider and your networking resources. Some (most?) providers know where to send the call based on registration. Each host can register as many accounts with as many providers as desired. Port forwarding can also solve a lot of problems. Switching to IAX (with registration) makes most of these issues moot. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chanspy on Asterisk 1.4.19
Guys,
Sean Bright wrote:
Steve Totaro wrote:
Should one have to change their dialplan for functionality to remain
the same in the same version?
I wasn't suggesting it wasn't a regression, just making the OP aware
that he can pass multiple arguments to a dialplan application (i.e.
ChanSpy(|bg(2000)))
He mentioned that he was able to get it to work in 1.4.19 by passing the
bridge argument ('b') but didn't seem to be aware that he could also
pass his original argument list ('g(2000)') as well. Seems easier to
just work around the problem with the additional argument than to
backport the application.
Yes I was aware of multiple arguments, I did try chanspy(|bg(2000)), I
tried all combinations I could think of. Although maybe what I should
have said was I tried chanspy(|b) just to prove chanspy itself was
working at all (and it was), with chanspy(|bg(2000)) the 'spygroup'
element didn't work, it just spied on every active call.
Anyway, I've raised a bug report as requested by Jared at Digium.
Steve
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[asterisk-users] End to end call monitoring?
We're having some difficulty tying together our Cisco and Audiocodes syslogs with our Trixbox asterisk logs. We'd like to have some way to split out a single call from all the activity going on at one moment. Obviously NTP is the first step for this, but we haven't found any means to tie the logs from the Cisco (which if we're lucky gives a UDP port number) and the Audiocodes (which rarely tell which of the 24 phone lines is generating the error or warning message). -HJC ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chanspy on Asterisk 1.4.19
My own Chanspy(g(GROUPNAME)) works 2 times out of three (roughly). The
other time, it crashes Asterisk. Using 1.4.19 too.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Rawlings
Sent: Thursday, April 17, 2008 14:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Chanspy on Asterisk 1.4.19
Guys,
Sean Bright wrote:
Steve Totaro wrote:
Should one have to change their dialplan for functionality
to remain
the same in the same version?
I wasn't suggesting it wasn't a regression, just making the
OP aware
that he can pass multiple arguments to a dialplan application (i.e.
ChanSpy(|bg(2000)))
He mentioned that he was able to get it to work in 1.4.19
by passing
the bridge argument ('b') but didn't seem to be aware that he could
also pass his original argument list ('g(2000)') as well. Seems
easier to just work around the problem with the additional argument
than to backport the application.
Yes I was aware of multiple arguments, I did try
chanspy(|bg(2000)), I tried all combinations I could think
of. Although maybe what I should have said was I tried
chanspy(|b) just to prove chanspy itself was working at all
(and it was), with chanspy(|bg(2000)) the 'spygroup'
element didn't work, it just spied on every active call.
Anyway, I've raised a bug report as requested by Jared at Digium.
Steve
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Re: [asterisk-users] Chanspy on Asterisk 1.4.19
Ah. My apologies for the confusion. Not that it helps you a great
deal, but I am running ChanSpy successfully in production (as we speak)
with 1.4.19 with no crashes or the like:
ChanSpy(SIP/11,g(Spyable))
Maybe its only a problem if no channel spec is passed?
Steve Rawlings wrote:
Guys,
Sean Bright wrote:
Steve Totaro wrote:
Should one have to change their dialplan for functionality to remain
the same in the same version?
I wasn't suggesting it wasn't a regression, just making the OP aware
that he can pass multiple arguments to a dialplan application (i.e.
ChanSpy(|bg(2000)))
He mentioned that he was able to get it to work in 1.4.19 by passing the
bridge argument ('b') but didn't seem to be aware that he could also
pass his original argument list ('g(2000)') as well. Seems easier to
just work around the problem with the additional argument than to
backport the application.
Yes I was aware of multiple arguments, I did try chanspy(|bg(2000)), I
tried all combinations I could think of. Although maybe what I should
have said was I tried chanspy(|b) just to prove chanspy itself was
working at all (and it was), with chanspy(|bg(2000)) the 'spygroup'
element didn't work, it just spied on every active call.
Anyway, I've raised a bug report as requested by Jared at Digium.
Steve
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--
Sean Bright
[EMAIL PROTECTED]
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Re: [asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
--- Nestor A. Diaz [EMAIL PROTECTED] wrote: ok, thanks, does rtp*timeout work if i have canreinvite=yes ? since rtp traffic is not passing thought asterisk, or i have to put canreinvite=no ? In my setup it doesn't really matter since calls are coming in through PSTN-IVR-QUEUE-SIP AGENT-TRANSFERS THROUGH ZAP PRI TO ANOTHER PBX. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
At 05:59 AM 4/17/2008, you wrote: Not at all little. If you have a lot of low priority outgoing traffic (i.e. p2p) saturating your link, uplink traffic shaping will mean the difference between a completely unintelligible call and something very acceptable. My network looks like this: Cable modem Linksys WRT54GS running Sveasoft LAN port 1 to the phone system running on it's own set of wires LAN ports 2-4 to everything else I've set the priority on port one to the highest and the priority on all the other ports to low and as far as I can tell, we've never had an issue where a big upload has impacted our voice calls. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] multiple users collisions
Hi, My dialplan works fine with one user (asking for the sharp key to be pressed to continue, and others), but when 2 users are calling at the same time if one press key # the two users are jumping to the next step. Anyidea ? FYI, I'm using Asterisk 1.4.10. -- Cyril SCETBON ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple users collisions
Logs? On Thu, Apr 17, 2008 at 11:47 PM, Cyril SCETBON [EMAIL PROTECTED] wrote: Hi, My dialplan works fine with one user (asking for the sharp key to be pressed to continue, and others), but when 2 users are calling at the same time if one press key # the two users are jumping to the next step. Anyidea ? FYI, I'm using Asterisk 1.4.10. -- Cyril SCETBON ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Moshe Brevda, CTO ipconnect, ltd. 26 Strauss St., Jerusalem, Israel W. 1.800.800.456 (+9722.569.5295) M. +97254.666.1367 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chanspy on Asterisk 1.4.19
Mike wrote:
My own Chanspy(g(GROUPNAME)) works 2 times out of three (roughly). The
other time, it crashes Asterisk. Using 1.4.19 too.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Rawlings
Sent: Thursday, April 17, 2008 14:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Chanspy on Asterisk 1.4.19
Guys,
Sean Bright wrote:
Steve Totaro wrote:
Should one have to change their dialplan for functionality
to remain
the same in the same version?
I wasn't suggesting it wasn't a regression, just making the
OP aware
that he can pass multiple arguments to a dialplan application (i.e.
ChanSpy(|bg(2000)))
He mentioned that he was able to get it to work in 1.4.19
by passing
the bridge argument ('b') but didn't seem to be aware that he could
also pass his original argument list ('g(2000)') as well. Seems
easier to just work around the problem with the additional argument
than to backport the application.
Yes I was aware of multiple arguments, I did try
chanspy(|bg(2000)), I tried all combinations I could think
of. Although maybe what I should have said was I tried
chanspy(|b) just to prove chanspy itself was working at all
(and it was), with chanspy(|bg(2000)) the 'spygroup'
element didn't work, it just spied on every active call.
Anyway, I've raised a bug report as requested by Jared at Digium.
Steve
This was an incredibly subtle bug that was introduced into 1.4.19 when the
other
work was done on chanspy to fix crashes and deadlocks. It has been fixed in 1.4
in SVN revision 114226.
Basically, chanspy was a crapshoot if you didn't specify a first argument,
because the function intended to walk through the list of active channels would
always end up returning the first channel it found. If that happened to be a
spy-able channel, then great, otherwise you'd never spy on anything.
Mark Michelson
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Re: [asterisk-users] CDR and transfers! :(
Raúl Gómez C. wrote:
Hi list,
snip
I think this is a very common scenario so, how are you doing to handle this
situation???
What if you were to set an account code to the extension that is
requesting the long-distance call?
So person at extension 111 requests a long distance call to
808-555-1212. Lets say the receptionist dials, then, *111*8085551212...
The PBX does something like:
exten = _*XXX*NXXNXX,1,SetAccountCode(${EXTEN:1:3})
exten = _*XXX*NXXNXX,n,Dial(Zap/G1/${EXTEN:5})
then, the receptionist transfers the call to extension 111, which again
sets the account code to111.
Seems the account code would help the CDRs to make more sense? Maybe I
overlooked something :)
When anybody in the office dials 111, however, the accountcode will
still be set in my scenario. This might lead to the user at 111 being
charged for inter-office calls! So:
exten = _XXX,1,Dial(SIP/${EXTEN})
exten = _#XXX,1,SetAccountCode(${EXTEN:1})
exten = _#XXX,2,Goto(${EXTEN:1},1)
And the receptionist transfers long distance calls to #111
Moj
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Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
J. Oquendo wrote: Its fine and dandy, but the problem is you're still getting 5 packets. You're still saturated period. No QoS in the world outside of your provider and more bandwidth can alleviate that. Your provider is not going to care what you do once its passed to the CPE. So look at it logically again. QoS on a home router... Useless COMING IN. Going out... Means little but helps MINIMALLY. I think the road to success, when talking about upstream at least, is partially paved by trying to keep maximum traffic at 4 packets instead of 5, if 5 is going to saturate the link. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
J. Oquendo wrote: it does, when someone can realistically point this out please let me know so I can switch from a DS3 to T1 and save money. Use the T1 for voice and get a DSL modem for your data use? :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Apparently, there is a SIP(diversionheader) field that fixes the problem below, but I cannot find any docs or examples of how to use it in my dialplan. Any help would be appreciated. We have a Cisco CallManager where users forward their numbers, so PSTN-PSTN calls get this error... -Greg --- SIP read from 209.253.136.204:5060 --- INVITE sip:[EMAIL PROTECTED];transport=UDP SIP/2.0 From: Cell Phone TXsip:[EMAIL PROTECTED];tag=501ff0a-13c4-4807c9fe-a5bac2-7f73fc7a To: CISTERA 9723814678sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Via: SIP/2.0/UDP 209.253.136.204:5060;branch=z9hG4bK-301-4807c9fe-a5bac2-5babdc2b Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY Supported: timer Accept: multipart/mixed,application/media_control+xml,application/sdp Max-Forwards: 9 Min-SE: 60 Contact: sip:[EMAIL PROTECTED]:5060;transport=UDP Content-Type: application/sdp Content-Length: 500 v=0 o=BroadWorks 31324769 1 IN IP4 209.253.136.204 s=- c=IN IP4 209.253.136.204 t=0 0 m=audio 24418 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=fmtp:101 0-15 a=fmtp:18 annexb=no a=x-cxc-sess:04c2e65cf9a2aa97-1 a=x-cxc-info:cGVlci1wdWI9NjQuMTk5LjUxLjIxMDtwZWVyLXNkcD0yMDkuMjUzLjEyOS4xNTc6MjYzMDY7 a=x-cxc-info:cGVlci1yb3V0ZS10YWc9aW50ZXJuYWw7YW5jaG9yLWRzdD0yMDkuMjUzLjEzNi4yMDQ6MjQ0MTg7 a=sendrecv - --- (14 headers 17 lines) --- Sending to 209.253.136.204 : 5060 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] Found peer 'McLeodUSA' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 209.253.136.204:24418 Found audio description format G729 for ID 18 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw| g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 209.253.136.204:24418 Looking for 9723814678 in default (domain 209.33.163.37) list_route: hop: sip:[EMAIL PROTECTED]:5060;transport=UDP ns2*CLI --- Transmitting (no NAT) to 209.253.136.204:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.253.136.204:5060;branch=z9hG4bK-301-4807c9fe-a5bac2-5babdc2b;received=209.253.136.204 From: Cell Phone TXsip:[EMAIL PROTECTED];tag=501ff0a-13c4-4807c9fe-a5bac2-7f73fc7a To: CISTERA 9723814678sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/4693412073-08fdbf78, SIP/[EMAIL PROTECTED]) in new stack Audio is at 192.168.5.14 port 13374 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.5.10:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.5.14:5060;branch=z9hG4bK4a44607b;rport From: Cell Phone TX sip:[EMAIL PROTECTED];tag=as178544f0 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 17 Apr 2008 22:08:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 240 v=0 o=root 28662 28662 IN IP4 192.168.5.14 s=session c=IN IP4 192.168.5.14 t=0 0 m=audio 13374 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called [EMAIL PROTECTED] ns2*CLI --- SIP read from 192.168.5.10:49365 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.5.14:5060;branch=z9hG4bK4a44607b;rport From: Cell Phone TX sip:[EMAIL PROTECTED];tag=as178544f0 To: sip:[EMAIL PROTECTED];tag=16863906 Date: Thu, 17 Apr 2008 22:06:54 GMT Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 - --- (9 headers 0 lines) --- ns2*CLI --- SIP read from 192.168.5.10:6060 --- INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.10:6060;branch=z9hG4bK32426484 From: Cell Phone TX sip:[EMAIL PROTECTED];tag=16863908 To: sip:[EMAIL PROTECTED] Date: Thu, 17 Apr 2008 22:06:55 GMT Call-ID: [EMAIL PROTECTED] Supported: timer Min-SE: 1800 User-Agent: Cisco-CCM4.1 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK CSeq: 101 INVITE Max-Forwards: 70 Remote-Party-ID: Cell Phone TX sip:[EMAIL PROTECTED];party=calling;screen=no;privacy=off Contact: sip:[EMAIL PROTECTED]:6060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 227 v=0 o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 192.168.5.10 s=SIP Call c=IN IP4
Re: [asterisk-users] End to end call monitoring?
On Thu, Apr 17, 2008 at 2:15 PM, Henry Cobb [EMAIL PROTECTED] wrote: We're having some difficulty tying together our Cisco and Audiocodes syslogs with our Trixbox asterisk logs. We'd like to have some way to split out a single call from all the activity going on at one moment. Obviously NTP is the first step for this, but we haven't found any means to tie the logs from the Cisco (which if we're lucky gives a UDP port number) and the Audiocodes (which rarely tell which of the 24 phone lines is generating the error or warning message). -HJC Can you pull the SIPCallID from the Cisco or Audiocodes? Those should match at least on each leg of the call. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR and transfers! :(
On Thu, Apr 17, 2008 at 5:14 PM, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
Raúl Gómez C. wrote:
Hi list,
snip
I think this is a very common scenario so, how are you doing to handle this
situation???
What if you were to set an account code to the extension that is
requesting the long-distance call?
So person at extension 111 requests a long distance call to
808-555-1212. Lets say the receptionist dials, then, *111*8085551212...
The PBX does something like:
exten = _*XXX*NXXNXX,1,SetAccountCode(${EXTEN:1:3})
exten = _*XXX*NXXNXX,n,Dial(Zap/G1/${EXTEN:5})
then, the receptionist transfers the call to extension 111, which again
sets the account code to111.
Seems the account code would help the CDRs to make more sense? Maybe I
overlooked something :)
When anybody in the office dials 111, however, the accountcode will
still be set in my scenario. This might lead to the user at 111 being
charged for inter-office calls! So:
exten = _XXX,1,Dial(SIP/${EXTEN})
exten = _#XXX,1,SetAccountCode(${EXTEN:1})
exten = _#XXX,2,Goto(${EXTEN:1},1)
And the receptionist transfers long distance calls to #111
Moj
Without putting too much thought into it, it would seem that
creatively using queues or maybe even a meetme room, you could get
accurate billing.
Maybe if you use some AGI/AMI mojo along with queue_log you could come
up with something solid? Not sure how using app_bridge shows up in
CDR either...
Just a thought.
Thanks,
Steve Totaro
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Re: [asterisk-users] Chanspy on Asterisk 1.4.19
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mark Michelson
Sent: Thursday, April 17, 2008 17:18
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Chanspy on Asterisk 1.4.19
Mike wrote:
My own Chanspy(g(GROUPNAME)) works 2 times out of three (roughly).
The other time, it crashes Asterisk. Using 1.4.19 too.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Steve
Rawlings
Sent: Thursday, April 17, 2008 14:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Chanspy on Asterisk 1.4.19
Guys,
Sean Bright wrote:
Steve Totaro wrote:
Should one have to change their dialplan for functionality
to remain
the same in the same version?
I wasn't suggesting it wasn't a regression, just making the
OP aware
that he can pass multiple arguments to a dialplan
application (i.e.
ChanSpy(|bg(2000)))
He mentioned that he was able to get it to work in 1.4.19
by passing
the bridge argument ('b') but didn't seem to be aware
that he could
also pass his original argument list ('g(2000)') as well. Seems
easier to just work around the problem with the
additional argument
than to backport the application.
Yes I was aware of multiple arguments, I did try
chanspy(|bg(2000)),
I tried all combinations I could think of. Although maybe what I
should have said was I tried
chanspy(|b) just to prove chanspy itself was working at
all (and it
was), with chanspy(|bg(2000)) the 'spygroup'
element didn't work, it just spied on every active call.
Anyway, I've raised a bug report as requested by Jared at Digium.
Steve
This was an incredibly subtle bug that was introduced into
1.4.19 when the other work was done on chanspy to fix crashes
and deadlocks. It has been fixed in 1.4 in SVN revision 114226.
Basically, chanspy was a crapshoot if you didn't specify a
first argument, because the function intended to walk through
the list of active channels would always end up returning the
first channel it found. If that happened to be a spy-able
channel, then great, otherwise you'd never spy on anything.
Mark Michelson
Mark,
I added a first argument. Here is my line now:
exten = *012,n,Chanspy(SIP,qg(GROUP_NAME))
Unfortunately, that still crashes Asterisk once out of 3-5 times. Is there
anyway to absolutely prevent crashes with this bug in vanilla 1.4.19?
Thanks,
Mike
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Re: [asterisk-users] Chanspy on Asterisk 1.4.19
I saw a patch attached to that bug report, just download it run patch
and then make clean make install, restart asterisk and you should be
smokin.
Mike wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mark Michelson
Sent: Thursday, April 17, 2008 17:18
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Chanspy on Asterisk 1.4.19
Mike wrote:
My own Chanspy(g(GROUPNAME)) works 2 times out of three (roughly).
The other time, it crashes Asterisk. Using 1.4.19 too.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Steve
Rawlings
Sent: Thursday, April 17, 2008 14:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Chanspy on Asterisk 1.4.19
Guys,
Sean Bright wrote:
Steve Totaro wrote:
Should one have to change their dialplan for functionality
to remain
the same in the same version?
I wasn't suggesting it wasn't a regression, just making the
OP aware
that he can pass multiple arguments to a dialplan
application (i.e.
ChanSpy(|bg(2000)))
He mentioned that he was able to get it to work in 1.4.19
by passing
the bridge argument ('b') but didn't seem to be aware
that he could
also pass his original argument list ('g(2000)') as well. Seems
easier to just work around the problem with the
additional argument
than to backport the application.
Yes I was aware of multiple arguments, I did try
chanspy(|bg(2000)),
I tried all combinations I could think of. Although maybe what I
should have said was I tried
chanspy(|b) just to prove chanspy itself was working at
all (and it
was), with chanspy(|bg(2000)) the 'spygroup'
element didn't work, it just spied on every active call.
Anyway, I've raised a bug report as requested by Jared at Digium.
Steve
This was an incredibly subtle bug that was introduced into
1.4.19 when the other work was done on chanspy to fix crashes
and deadlocks. It has been fixed in 1.4 in SVN revision 114226.
Basically, chanspy was a crapshoot if you didn't specify a
first argument, because the function intended to walk through
the list of active channels would always end up returning the
first channel it found. If that happened to be a spy-able
channel, then great, otherwise you'd never spy on anything.
Mark Michelson
Mark,
I added a first argument. Here is my line now:
exten = *012,n,Chanspy(SIP,qg(GROUP_NAME))
Unfortunately, that still crashes Asterisk once out of 3-5 times. Is there
anyway to absolutely prevent crashes with this bug in vanilla 1.4.19?
Thanks,
Mike
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--
Thank you and have any kind of day you want,
Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]
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[asterisk-users] SIP outboundproxy for asterisk
Hi, I have searched through the archives on this mailing list, but didn't find a solution to the outboundproxy problem. Can someone please help? I wish to configure Asterisk such that all outgoing SIP requests get relayed to an outboundproxy, instead of the actual recipient directly. In my setup, I wish to use OpenSER as an outboundproxy that is running on the same box and network-interface as Asterisk itself, but listening on a different SIP port. I have tried the following, but none has worked: 1) Specify an outboundproxy in sip.conf [general] Outboundproxy=10.1.1.102 Asterisk still goes out to the callee directly. 2) I tried the following as well: exten = _.,n,Dial(SIP/10.1.1.102/[EMAIL PROTECTED]) In this case OpenSER receives an INVITE with a RURI as - INVITE 10.1.1.102/[EMAIL PROTECTED] Which obviously is not successfully parsed by OpenSER. Can you please help me configure Asterisk to use my OpenSER as an Outboundproxy for all outgoing call legs? Thanks in advance, Regards, Amit. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP outboundproxy for asterisk
There are lots of different ways to configure Asterisk and SER to get them working together depending on what you want to do. The link below is not a bad starting point. http://www.voip-info.org/wiki-Asterisk+at+large Asterisk has outboundproxy and outboundproxyport settings that can be used in sip.conf but while they work to an extent there are bugs with using them that have been around for at least two years and in fact as fas as I can tell the outboundproxyport setting is ignored even though it gets parsed. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
