[asterisk-users] Configuring 2 Asterisk servers with a SIP trunk
Hi All, Please let me know the how to configure a SIP trunk of a asterisk Server with another one (not IAX2). Asterisk-A should register a SIP trunk with Asterisk-B server . With Regards Alok Ranjan Mohapatra Software Engineer +91 9866269992 PrimeSoft IP Solutions (P) Ltd # 917- 922,East Wing, 9th floor Block III, White House,Begumpet Hyderabad - 500016, INDIA Ph - 91-40-23418239/40 www.primesoftindia.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 501 + Voicemail notification
Hi, I was wondering if anyone was aware of a work around/fix for the following Polycom + Asterisk (1.2.7.1) bug, if this was doable in Asterisk, or if it is just a bug in the Polycom firmware. (Appears in both the 1.4.x release and the 1.6.7 releases) The SIP message for voicemail notify on a new message reports the correct amounts (let's say 1 new, 3 old) The phone will display new 1, and old 3 Then when the message becomes read (ie old), a new SIP notify message gets sent to the phone (0 new, 4 old), but the phone reports 0 new, and 0 old messages. -- Gerwin van de Steeg Engineer Vadacom Ltd W: www.vadacom.co.nz E: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN-BRI issue
Tzafrir Cohen ha scritto: Works fine with Junghanns' cards. One simple thing for you to test: set one port in TE mode and one port in NT mode (move all 5 jumbers of that port to the other position to get it into NT mode). Then try to make a loopback connection (using a standard ethernet cable). Here you control both ends and thus there are less configuration pains. One thing that could be wrong is if both sides do not agree on the line settings. Where do you connect to? What do you have on zaptel.conf ? On zapata.conf? The very first test I did was to set a loopback between two spans (one in nt mode, the other in te mode). I wrote a small script to setup calls between the span continuously, and... guess what? I let the system run for 4 hours, place about 20.000 calls with no problem at all. Unforutnately, as soon as I connect the spans in te mode to Telecom Italia's NT1 lines, after a random time (from 15 minutes to 1 hour) and a random number of outgoing calls (the first 30 calls gets routed with no problem), one or more spans begin reporting Layer 1 Down. I tried to do a bri intense debug span x, all I see are SABME packets sent from the card to the NT1 line, with no UA reply, but INCOMING CALLS ARE WORKING ANYWAY!! The most strange thing at all is to be on the phone speaking with someone, on the very same line which bristuff is complaining about, in the very same moment in which bristuff is reporting layer 1 down... This does not happen only on my line. I saw it with my own eyes on 3 different asterisk boxes with 3 different Junghanns quadBRI, on 3 different PC (IBM/HP), on 3 different sets of ISDN lines (p2p, p2mp...) Just fyi, here's my really simple zaptel/zapata config (btw, I think I've tried all n! permutations of parameters for over two weeks...): bristuff 0.3.0-pre-1v, asterisk 1.2.13, zaptel 1.2.10, libpri 1.2.4 zaptel: loadzone=it defaultzone=it span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,1,3,ccs,ami bchan=4-5 dchan=6 span=3,1,3,ccs,ami bchan=7-8 dchan=9 zapata: language=it switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown resetinterval=never priindication=outofband callprogress=yes usecallingpres=yes echocancel=yes echocancelwhenbridged=yes echotraining=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 context=ingresso signalling=bri_cpe channel = 1-2 channel = 4-5 signalling=bri_cpe_ptmp channel = 7-8 -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] wi-fi ip phone scenario
Andrew Joakimsen ha scritto: Are you using WDS? While it won't totally fix every issue, I've found in my trials that turning off WDS and making sure all the AP were connected to the same wired network was way more reliable, no more random unregistartion and issue with registering (still seems to unregister at times, but re-registartion won't require a reboot). Nope. I've a group of 12 indoor colubris access points + 2 outdoor ones, and a msc5200 controller unit (the area is quite wide with two 4-storey buildings, two 8 sq.feet hangars, one 4 sq.feet outdoor parking lot) no wds, all of them are wired to ethernet switches, no wireless bridging. Diversity on all APs, automatic transimt power and channel selection (that seems to work when monitoring devices). I've tried linksys wip300/wip330, nokia e60/e70, utstarcom f1000/f1000g, samsung wip6000. The main problem (apart from firmware bugs/crashes which I hope should be fixed on newer versions) is that phones tend to stick to their associated AP even when it's clearly time to move to the next AP: if you watch the phone's rssi indicator while you walk inside the coverage area, you can see the value decreasing to almost no signal before reassociating, which is unacceptable. The only phones which seem to deal very well with it are the nokia eSeries. Unfortunately they have many other major issues. (two weeks ago I gave 3 firmware-updated e60 to my bosses to replace their cellphones and after one week they almost threw them back to me, complaining about the fact they had to reboot the phone at least twice a day because it freezed...) Also, on asterisk the qualify=2000 sip setting seems to be to low, as the console shows repeated LAGGED/UNREACHABLE/REACHABLE notices on those phones. From this experience I think wi-fi technology is not really mature enough to replace dect right now. Maybe I'm wrong. At least I hope so, my bosses are not really happy with the new system. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2/SIP Wifi Phones
Hello All, I apologize beforehand if this is of topic. I am looking for a solution for Wifi phones to access asterisk via iax2 or sip. The trouble is that it is on a manufacturing shop floor, so reception and noise cancellation is going to be a concern. Are there any non 2.4Ghz solutions for Asterisk? Probably in the 900Mhz spectrum. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] translate.c:88 powerof: Powerof 0: No power??
Hi,I have just installed fresh FreeBSD 6.1 and asterisk. But I get warning about power. I have read forums that others had the same problems, but there is no solution how to fix this problem. Maybe somebody knows how to fix this problem... This is my machine configurations:KERNEL:options IPFIREWALLoptions DUMMYNEToptions HZ=1000and commented these:#cpu I486_CPU#cpu I586_CPU #options INET6I have installed asterisk from ports. This is my installed programs for asterisk:mpg123-0.59r_17asterisk-1.2.13asterisk-addons-1.2.3_1spandsp-0.0.2.p26zaptel-1.0_1libpri-1.2.3 Loaded modules:Id Refs Address Size Name1 9 0xc040 66a2a4 kernel2 1 0xc0a6b000 58554 acpi.ko3 2 0xc367f000 31000 zaptel.ko4 1 0xc36b4000 2000 ztdummy.ko 5 1 0xc36cd000 16000 linux.ko Warnings:Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power??Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power??Oct 28 16:38:35 WARNING[429]: translate.c:133 ast_translator_build_path: No translator path from unknown to unknown Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power??Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power??Oct 28 16:38:35 WARNING[429]: translate.c:133 ast_translator_build_path: No translator path from unknown to unknown Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power??Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power??Oct 28 16:38:35 WARNING[429]: translate.c:133 ast_translator_build_path: No translator path from unknown to unknown Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power??Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power??Oct 28 16:38:35 WARNING[429]: translate.c:133 ast_translator_build_path: No translator path from unknown to unknown Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power??Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power??Oct 28 16:38:35 WARNING[429]: translate.c:133 ast_translator_build_path: No translator path from unknown to unknown Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power??Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power??Oct 28 16:38:35 WARNING[429]: translate.c:133 ast_translator_build_path: No translator path from unknown to unknown ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: SIP v IAX2
On 27 Oct 2006, at 16:42, Roberto Pereyra wrote: Hi Which is most resistant to the loss of packages in a dirty link ? SIP or IAX ? Well there isn't much in it at the protocol level, at least in terms of the odd packet being lost. IAX does have an advantage when it comes to the link dropping completely. IAX will detect total packet loss (in either direction) and hang up the call after about 30 seconds. SIP can often not notice the RTP stream has died (esp if the control channel is still up) leaving you with half a call hanging around. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: 0 channels configured with tdm400 (tdm04b rev. G)
More info. [EMAIL PROTECTED] ~]# ztcfg -vv Zaptel Configuration == Channel map: 0 channels configured. cat /etc/modprobe.conf alias scsi_hostadapter ahci alias usb-controller ehci-hcd alias usb-controller1 uhci-hcd alias eth0 e100 alias eth1 3c59x alias wcfxs wctdm install wctdm /sbin/modprobe --ignore-install wctdm /sbin/ztcfg *** [EMAIL PROTECTED] ~]# cat /etc/asterisk/zaptel.conf fxsls=1-4 loadzone=us defaultzone=us *** [EMAIL PROTECTED] ~]# cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 1 WCTDM/0/0 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 [EMAIL PROTECTED] ~]# On 10/27/06, Erick Perez [EMAIL PROTECTED] wrote: Hi, Some weird problem (or I'm too sleepy) happens with a tdm04B revision G (4fxo) Steps: modprobe zaptel modprobe wctdm ztcfg -vv /etc/zaptel.conf fxsls=1-4 # TDM04B defaultzone=us loadzone=us /etc/asterisk/zapata.conf signalling=fxs_ls group=1 context=incoming channel = 1-4 modprobe zaptel and wctdm load fine, however ztcfg -vv shows: 0 channels configured Im using centos 4.4 with Asterisk Version 1.2.13 Zaptel Version 1.2.10 Libpri Version 1.2.4 Physically looking at the card, the four FXO ports have the green led turned on. It has no IRQ conflicts and zaptel compiled cleanly. Kernel is 2.6.9-42ELsmp (it's a dual core Intel machine in an Intel 945G board) Your comments are welcomed. -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: 0 channels configured with tdm400 (tdm04b rev. G)
Tzafrir, please disregard my previous postdefinitely i was WAAAY sleepy. The error was simple (now that I just waked up completely): I was touching /etc/asterisk/zaptel.conf instead of /etc/zaptel.conf. I should remind myself not to work past midnight. I don't recall what made me think the file was inside the asterisk directory. Thanks and apologies to all,. On 10/28/06, Erick Perez [EMAIL PROTECTED] wrote: More info. [EMAIL PROTECTED] ~]# ztcfg -vv Zaptel Configuration == Channel map: 0 channels configured. cat /etc/modprobe.conf alias scsi_hostadapter ahci alias usb-controller ehci-hcd alias usb-controller1 uhci-hcd alias eth0 e100 alias eth1 3c59x alias wcfxs wctdm install wctdm /sbin/modprobe --ignore-install wctdm /sbin/ztcfg *** [EMAIL PROTECTED] ~]# cat /etc/asterisk/zaptel.conf fxsls=1-4 loadzone=us defaultzone=us *** [EMAIL PROTECTED] ~]# cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 1 WCTDM/0/0 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 [EMAIL PROTECTED] ~]# On 10/27/06, Erick Perez [EMAIL PROTECTED] wrote: Hi, Some weird problem (or I'm too sleepy) happens with a tdm04B revision G (4fxo) Steps: modprobe zaptel modprobe wctdm ztcfg -vv /etc/zaptel.conf fxsls=1-4 # TDM04B defaultzone=us loadzone=us /etc/asterisk/zapata.conf signalling=fxs_ls group=1 context=incoming channel = 1-4 modprobe zaptel and wctdm load fine, however ztcfg -vv shows: 0 channels configured Im using centos 4.4 with Asterisk Version 1.2.13 Zaptel Version 1.2.10 Libpri Version 1.2.4 Physically looking at the card, the four FXO ports have the green led turned on. It has no IRQ conflicts and zaptel compiled cleanly. Kernel is 2.6.9-42ELsmp (it's a dual core Intel machine in an Intel 945G board) Your comments are welcomed. -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is it possible to connect two servers using SIP?
Hi Friends,I have created SIP extensions in our two Asterisk servers. Now, I want to connect these two servers using SIP. I searched a lot in internet about this. But, I found that there is a possibility to connect two servers using IAX2 only. Is it possible to connect two Asterisk servers using SIP? If so, can you give me a tutorial link about this?Looking forward to your response. Thank you.Regards,Chandra. We have the perfect Group for you. Check out the handy changes to Yahoo! Groups. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: 0 channels configured with tdm400 (tdm04b rev. G)
On Sat, Oct 28, 2006 at 09:24:52AM -0500, Erick Perez wrote: More info. [EMAIL PROTECTED] ~]# ztcfg -vv Zaptel Configuration == Channel map: ztcfg has heard of no channels, and hence did not try to configure them. Why is that? 0 channels configured. cat /etc/modprobe.conf alias scsi_hostadapter ahci alias usb-controller ehci-hcd alias usb-controller1 uhci-hcd alias eth0 e100 alias eth1 3c59x alias wcfxs wctdm install wctdm /sbin/modprobe --ignore-install wctdm /sbin/ztcfg *** [EMAIL PROTECTED] ~]# cat /etc/asterisk/zaptel.conf ... Because ztcfg looks for /etc/zaptel.conf and not for /etc/asterisk/zaptel.conf . fxsls=1-4 loadzone=us defaultzone=us *** [EMAIL PROTECTED] ~]# cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 1 WCTDM/0/0 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 [EMAIL PROTECTED] ~]# -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] translate.c:88 powerof: Powerof 0: No power??
This is a problem of the codec you are attempting to use. Wich codec is?, it seems to Asterisk cannot identify the codec you are using. Regards On 10/28/06, Giedrius Augys [EMAIL PROTECTED] wrote: Hi, I have just installed fresh FreeBSD 6.1 and asterisk. But I get warning about power. I have read forums that others had the same problems, but there is no solution how to fix this problem. Maybe somebody knows how to fix this problem... This is my machine configurations: KERNEL: options IPFIREWALL options DUMMYNET options HZ=1000 and commented these: #cpuI486_CPU #cpuI586_CPU #optionsINET6 I have installed asterisk from ports. This is my installed programs for asterisk: mpg123-0.59r_17 asterisk-1.2.13 asterisk-addons-1.2.3_1 spandsp-0.0.2.p26 zaptel-1.0_1 libpri-1.2.3 Loaded modules: Id Refs AddressSize Name 19 0xc040 66a2a4 kernel 21 0xc0a6b000 58554acpi.ko 32 0xc367f000 31000zaptel.ko 41 0xc36b4000 2000 ztdummy.ko 51 0xc36cd000 16000linux.ko Warnings: Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:133 ast_translator_build_path: No translator path from unknown to unknown Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:133 ast_translator_build_path: No translator path from unknown to unknown Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:133 ast_translator_build_path: No translator path from unknown to unknown Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:133 ast_translator_build_path: No translator path from unknown to unknown Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:133 ast_translator_build_path: No translator path from unknown to unknown Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:133 ast_translator_build_path: No translator path from unknown to unknown ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zap disconnect
Hi List, Im having a bit of an odd problem with asterisk and outgoing zap calls. Tzafrir has been kind enough to help me get the logging sorted out so I have some idea of whats going wrong, but Im a little flummoxed. Essentially the symptoms are as follows; Make a SIP call from Cisco 7960 or 7940 to asterisk, where it is routed out on a ZAP (x100p) line. After exactly 2mins 40seconds (+- 1sec) the call is hung up, normal clearing (although obviously its actually an analogue line) The same symptoms are not experienced with incoming calls on the same zap line, nor with internal SIPSIP calls, or external SIPSIP or SIPIAX calls. With full logging enabled, just before the hang up, this is written to the log: Oct 26 18:04:59 DEBUG[10822] chan_sip.c: = No match Their Call ID: 5907c40463e18 [EMAIL PROTECTED] Their Tag as6e46885a Our tag: as2b18a370 Oct 26 18:04:59 DEBUG[10822] chan_sip.c: = Found Their Call ID: 003094c4-4ea9000 [EMAIL PROTECTED] Their Tag 003094c44ea90008745cabf1-6f4660c7 Our tag : as56514c99 Oct 26 18:04:59 DEBUG[10822] chan_sip.c: Received CANCEL (14) - Command in SIP CANCEL Oct 26 18:04:59 DEBUG[10865] channel.c: Hanging up channel 'Zap/2-1' Oct 26 18:04:59 DEBUG[10865] chan_zap.c: zt_hangup(Zap/2-1) Oct 26 18:04:59 DEBUG[10865] chan_zap.c: Hangup: channel: 2 index = 0, normal = 21, callwait = -1, thirdcall = -1 Which seems to indicate that the phone actually issued a cancel (of course, unless Im reading it wrong). Has anyone seen this before? Or had any ideas of what it could be? Or even just where to start looking Very best regards, Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now
BTW: as an alternative to zaphfc+flotz, consider vzaphfc. It seems that the only place from which you can download an up-to-date version nowadays is the Debian zaptel package: http://svn.debian.org/wsvn/pkg-voip/zaptel/trunk/vzaphfc/ http://packages.debian.org/zaptel-source Thanks! I tried looking for some more info on the Debian pages about what vzaphfc exactly is, but couldn't find any documentation of it. Is there any main page? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] [OT] wi-fi ip phone scenario
Alberto, you should have bought a dect solution, the dect technology is far better at swapping between cells. Wifi is still a little immature at this time. Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Pastore Sent: Saturday, 28 October 2006 6:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [OT] wi-fi ip phone scenario Andrew Joakimsen ha scritto: Are you using WDS? While it won't totally fix every issue, I've found in my trials that turning off WDS and making sure all the AP were connected to the same wired network was way more reliable, no more random unregistartion and issue with registering (still seems to unregister at times, but re-registartion won't require a reboot). Nope. I've a group of 12 indoor colubris access points + 2 outdoor ones, and a msc5200 controller unit (the area is quite wide with two 4-storey buildings, two 8 sq.feet hangars, one 4 sq.feet outdoor parking lot) no wds, all of them are wired to ethernet switches, no wireless bridging. Diversity on all APs, automatic transimt power and channel selection (that seems to work when monitoring devices). I've tried linksys wip300/wip330, nokia e60/e70, utstarcom f1000/f1000g, samsung wip6000. The main problem (apart from firmware bugs/crashes which I hope should be fixed on newer versions) is that phones tend to stick to their associated AP even when it's clearly time to move to the next AP: if you watch the phone's rssi indicator while you walk inside the coverage area, you can see the value decreasing to almost no signal before reassociating, which is unacceptable. The only phones which seem to deal very well with it are the nokia eSeries. Unfortunately they have many other major issues. (two weeks ago I gave 3 firmware-updated e60 to my bosses to replace their cellphones and after one week they almost threw them back to me, complaining about the fact they had to reboot the phone at least twice a day because it freezed...) Also, on asterisk the qualify=2000 sip setting seems to be to low, as the console shows repeated LAGGED/UNREACHABLE/REACHABLE notices on those phones. From this experience I think wi-fi technology is not really mature enough to replace dect right now. Maybe I'm wrong. At least I hope so, my bosses are not really happy with the new system. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now
On Sat, Oct 28, 2006 at 05:17:02PM +0200, Remco Barendse wrote: BTW: as an alternative to zaphfc+flotz, consider vzaphfc. It seems that the only place from which you can download an up-to-date version nowadays is the Debian zaptel package: http://svn.debian.org/wsvn/pkg-voip/zaptel/trunk/vzaphfc/ http://packages.debian.org/zaptel-source Thanks! I tried looking for some more info on the Debian pages about what vzaphfc exactly is, but couldn't find any documentation of it. Is there any main page? As I sad before: no. Actually there is, but it contains much dated version of vzaphfc. Please contact the author for more information. I stress again that I have never tried it as I don't have a HFC-s card. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asteroid SIP Denial of Service Tool
Asteroid is a SIP denial of service attack tools which affected older versions of Asterisk the Open Source PBX and may affect other products running the SIP protocol. There are thousands of custom (mis)crafted SIP packets which were sent to a older versions of Asterisk that caused errors stopping Asterisk. The packets were crafted based on packetdumps from Wireshark with flags set for pseudo-spoofing, ranDUMBized extensions, etc.. The purpose of the tool was to help me understand SIP security and Denials of Service attacks on the SIP protocol. Originally I had intended on testing out my nCite Session Border Controller but after watching nCite crash and burn on its own, it made little sense for me to point it at it. I have found that by sending a certain sequence of these packets, in a certain order, servers react differently. Sometimes it crashed faster, sometimes more extensions subscribed, sometimes voicemails were created and the list went on. Asterisk version 1.2.13 and better are now patched from this issue but there are other products it has not been tested on. The packets were butchered in Perl and called from a shell script since I had to manipulate packet sequences individually. This Proof of Concept program is released to the public under the hopes that individuals will find a useful purpose for assessing DoS vulnerabilities. It is unfortunate though that there are idiots who will use this lame tool for malicious purposes. Some vendors, CERT and other organizations were contacted as early as September 9th 2006 to address issues with their products. Most reacted quickly to get the fixes in order. Thanks to Kevin P. Flemming and the guys on Asterisk Dev for creating a thread on this. Dan York for getting some to pay attention. PSIRT at Cisco for looking into this, Tim Donahue for his perl pointers, vgersh99 (aka vlad) for nawk foo pointers, PHV, Annihilannic, p5wizard (segment!), and Henning Schulzrinne for taking a look at the tool during his seminars at Columbia. Also thanks to Anthony LaMantia, Tzafir Cohen, and the others on the dev list for tolerating my posts. Public apologies to Jay R. Ashworth for my mis-reading of the (Missed)Trust in Caller ID thread on VOIPSA ;) Coming 10/31/2006 http://www.infiltrated.net/asteroid/ -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo echo @infiltrated|sed 's/^/sil/g;s/$/.net/g' http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 How a man plays the game shows something of his character - how he loses shows all - Mr. Luckey ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [OT] wi-fi ip phone scenario
On 2006-10-28 03:51:57 -0700, Alberto Pastore [EMAIL PROTECTED] said: Andrew Joakimsen ha scritto: Are you using WDS? While it won't totally fix every issue, I've found in my trials that turning off WDS and making sure all the AP were connected to the same wired network was way more reliable, no more random unregistartion and issue with registering (still seems to unregister at times, but re-registartion won't require a reboot). Nope. I've a group of 12 indoor colubris access points + 2 outdoor ones, and a msc5200 controller unit (the area is quite wide with two 4-storey buildings, two 8 sq.feet hangars, one 4 sq.feet outdoor parking lot) no wds, all of them are wired to ethernet switches, no wireless bridging. Diversity on all APs, automatic transimt power and channel selection (that seems to work when monitoring devices). I've tried linksys wip300/wip330, nokia e60/e70, utstarcom f1000/f1000g, samsung wip6000. The main problem (apart from firmware bugs/crashes which I hope should be fixed on newer versions) is that phones tend to stick to their associated AP even when it's clearly time to move to the next AP: if you watch the phone's rssi indicator while you walk inside the coverage area, you can see the value decreasing to almost no signal before reassociating, which is unacceptable. The only phones which seem to deal very well with it are the nokia eSeries. Unfortunately they have many other major issues. (two weeks ago I gave 3 firmware-updated e60 to my bosses to replace their cellphones and after one week they almost threw them back to me, complaining about the fact they had to reboot the phone at least twice a day because it freezed...) Also, on asterisk the qualify=2000 sip setting seems to be to low, as the console shows repeated LAGGED/UNREACHABLE/REACHABLE notices on those phones. From this experience I think wi-fi technology is not really mature enough to replace dect right now. Maybe I'm wrong. At least I hope so, my bosses are not really happy with the new system. If you set them all to the same channel (ie not automatic) it will work. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] tx_fax not getting entire fax
Steve, I am trying to get tx_fax to work. I am using a TDM2401E card. I have a 3 page fax and I only receive the first page on every attempt. I think I have enabled debug output below. Can you tell me what the problem might be? I am using snapshot from oct 26. asterisk 1.2.13 and libtiff 3.6.1-12 from redat/centos 4.4. THanks, Jerry - Oct 28 13:13:40 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Set rx type 4 Oct 28 13:13:40 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Set tx type 0 Oct 28 13:13:42 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Set rx type 0 Oct 28 13:13:42 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Set tx type 8 Oct 28 13:13:43 DEBUG[22746]: chan_sip.c:1412 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Oct 28 13:13:43 DEBUG[22738]: pbx_spool.c:319 scan_service: Delaying retry since we're currently running '?? x(?' Oct 28 13:13:43 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Set rx type 0 Oct 28 13:13:43 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Set tx type 4 Oct 28 13:13:44 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Set rx type 4 Oct 28 13:13:44 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Set tx type 0 Oct 28 13:13:46 DEBUG[22738]: pbx_spool.c:319 scan_service: Delaying retry since we're currently running '?? x(?' Oct 28 13:13:46 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Set rx type 0 Oct 28 13:13:46 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Set tx type 1 Oct 28 13:13:46 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Set rx type 13 Oct 28 13:13:46 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX FAX exchange complete Oct 28 13:13:46 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Set tx type 13 Oct 28 13:13:46 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX FAX exchange complete Oct 28 13:13:48 DEBUG[22745]: chan_iax2.c:7074 socket_read: Immediately destroying 3, having received hangup Oct 28 13:13:48 DEBUG[22763]: app_txfax.c:256 txfax_exec: Got hangup Oct 28 13:13:48 DEBUG[22763]: pbx.c:2341 __ast_pbx_run: Extension smvoice_faxout, priority 1 returned normally even though call was hung up ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [OT] wi-fi ip phone scenario
On 2006-10-27 11:55:14 -0700, Andrew Joakimsen [EMAIL PROTECTED] said: Are you using WDS? While it won't totally fix every issue, I've found in my trials that turning off WDS and making sure all the AP were connected to the same wired network was way more reliable, no more random unregistartion and issue with registering (still seems to unregister at times, but re-registartion won't require a reboot). I think it's cleary true that wiring WIFI infrastructure is easier and more reliable then WDS. On the other hand, I have been running my little network with WDS for over three weeks now, and it has been completely reliable. The tricks where to configure things properly and to have the bases closer together then one would think would be needed. Once this was setup. It works, and it keeps working. We had a couple of stress tests also, one black out and one unplugged router (carpenter). Came up cleanly and continued working fine. No mis-registrations and no problems. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [OT] wi-fi ip phone scenario
On 2006-10-28 07:55:43 -0700, Dean Collins [EMAIL PROTECTED] said: Alberto, you should have bought a dect solution, the dect technology is far better at swapping between cells. Wifi is still a little immature at this time. Not if correctly configured. This is simply wrong. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Diva server 4bri and Portuguese BRI
Hello, I need to connect one diva server 4bri to a portuguese BRI interface. The operator (PT) said that this bri is in point-to-multipoint mode (S0). Previously one PBX has connected to that interface. The asterisk and diva drivers are working ok but i cannot communicate to outside via this bri. Xlite gives me the message: call failed: declined. Anyone have experience with this setup? What are the main parameters for bri card configuration? D-channel protocol: ETSI-DSS1 or other? Interface mode: NT or TE? Direct Inward Dialing (DID): Yes ou no? (MSN ou DID?) Thanks by any kind of help! Best regards, PS. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Would you support a Bristuff mailing list ?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Olivier wrote: Hi, It seems to me that Bristuff usage has reached a point which implies a dedicated mailing list. This list would be of major use for : - bugs assessment - features requests - comments on Asterisk news Who seconds that ? Would it be difficult to make this happen ? Regards In general, I like the idea, but... Would enough people subscribe to it to make it work? Where would it be hosted? - assuming you use mailman, there would need to be a web/mail server, 1 or more sysadmins to administer the list (ever been the victim of a mail loop?). This will require a not insignificant amount of somebodies time and bandwidth. Not that I am trying to put you off the idea, but it does need to be considered very carefully. - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.137621 Linux Counter No. 202120 FWD:519961 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEVAwUBRUOzoktP/KMNOfRbAQI6lggAspl0JO5c9lKAQqbxwZL0IC+T5IOkurJX A8cwO1LYH49t8OcI8rQhrtd8QDw1nfdGuROz4g8t2oqyp9/12m3vu6BzbCYoBV9C 2Pu+dx9E1HCKIfnpbbMc0SaOrzMACm9AMa9m3wM98L131H6GAvU+TthejqA9KWfv PuY5K0NuYp/EnebjCP/CAznHaogBhBm5K1eUES8sHa8MuSkMhNLL6QZAt96Eq90G D/cNG44piV6KgxXH50nkW4nlbmSGNN73M0dxn/p+AA2tV2M3El/SLgZCK1NtLHbk zCcap+wAqXdcLAg696RKH2g5UGqJ+JaS6Szxss/8bV8Q2R5ZltfiSw== =gNxe -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Would you support a Bristuff mailing list ?
On 20:46, Sat 28 Oct 06, Ron Wellsted wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Olivier wrote: Hi, It seems to me that Bristuff usage has reached a point which implies a dedicated mailing list. This list would be of major use for : - bugs assessment - features requests - comments on Asterisk news Who seconds that ? Would it be difficult to make this happen ? Regards In general, I like the idea, but... Would enough people subscribe to it to make it work? Where would it be hosted? - assuming you use mailman, there would need to be a web/mail server, 1 or more sysadmins to administer the list (ever been the victim of a mail loop?). This will require a not insignificant amount of somebodies time and bandwidth. Not that I am trying to put you off the idea, but it does need to be considered very carefully. In general junghanns.net should host it. But looking at how active they are I dont see this happen. If enough people want a list like this I can setup one. I have both the software running for a bunch of other lists and I have unmetered bandwidth. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queues: roundrobin w/ reset (circular call distribution)
It looks like this issue has been raised before, but I see it mostly ignored and no answers given, so here it is again: For the past couple of years, I've wanted a queue that works very similarly to roundrobin/rrmemory, but that doesn't remember anything about where the last ring went to. This new strategy would always start at the first member (as defined in queues.conf) when a new call came in. I actually coded up a simple version of this (that I've been using for a couple of years) which I called roundrobinreset. The only problem is I coded that for 0.9.0 and am now moving to 1.2.13 and would like to not have to port that code every time I move versions. I'd be willing to port it one more time to this version if there's a chance of it getting integrated. Alternatively, am I missing some method through the current features where I could accomplish this? There are references to being able to implement circular call distribution (which is the unofficial name for what I'm talking about, I think) here: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+queues.conf and in mailing list posts, but I've tried that using the penalty approach and it doesn't seem to work. As other users have noted, the call just hangs on the lowest penalty member and never moves off of it. So what's up w/ this? What approach should I follow? Adding another queue strategy? Thanks, John Lawler ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk behind NAT and without portforwarding for rtp
Hi, I have an Asterisk behind NAT. NAT=yes and canreinvite=no in globals and for the peer. I call an peer. The peer advice to use another IP for the audio and my Asterisk is sending audio stream to the Audio server. Because of missing port forwarding I will not receive the audio stream and hear nothing. I would expect that Asterisk will cancel the connection, but this didnt happened. Asterisk will follow the reinvite from the peer. Any solution except portforwarding? best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind NAT and without portforwarding forrtp
yup. use IAX - Original Message - From: Thomas Winter [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, October 29, 2006 1:26 AM Subject: [asterisk-users] Asterisk behind NAT and without portforwarding forrtp Hi, I have an Asterisk behind NAT. NAT=yes and canreinvite=no in globals and for the peer. I call an peer. The peer advice to use another IP for the audio and my Asterisk is sending audio stream to the Audio server. Because of missing port forwarding I will not receive the audio stream and hear nothing. I would expect that Asterisk will cancel the connection, but this didnt happened. Asterisk will follow the reinvite from the peer. Any solution except portforwarding? best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind NAT and without portforwarding forrtp
Half asleep. Sorry for my last post. I believe you still need port forwarding for IAX. Time to keep to my bed time. - Original Message - From: Thomas Winter [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, October 29, 2006 1:26 AM Subject: [asterisk-users] Asterisk behind NAT and without portforwarding forrtp Hi, I have an Asterisk behind NAT. NAT=yes and canreinvite=no in globals and for the peer. I call an peer. The peer advice to use another IP for the audio and my Asterisk is sending audio stream to the Audio server. Because of missing port forwarding I will not receive the audio stream and hear nothing. I would expect that Asterisk will cancel the connection, but this didnt happened. Asterisk will follow the reinvite from the peer. Any solution except portforwarding? best regards Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Compiling Zaptel 1.2.10 on Ubuntu 6.10
Here's a weird problem that I'm not quite sure how to resolve. Zaptel 1.2.10 compiles just fine with make, but when make install is run, this happens: [ `id -u` = 0 ] /sbin/depmod -a 2.6.17-10-generic || : [ -f /etc/zaptel.conf ] || install -D -m 644 zaptel.conf.sample /etc/zaptel.conf build_tools/genmodconf linux26 tor2 torisa wcusb wcfxo wctdm wctdm24xxp ztdynamic ztd-eth wct1xxp wcte11xp pciradio ztd-loc ztdummy [: 66: ==: unexpected operator [: 66: ==: unexpected operator Unknown kernel build version requested... exiting. make: *** [install] Error 1 This worked just fine under ubuntu 6.06 with the same set of packages installed. Any help is appreciated. -- Strom Carlson http://www.stromcarlson.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoIP GSM Gateways
Im looking at setting up a VoIP GSM gateway to connect to my asterisk box. What experience have people on this list have with GSM gateway hardware. I have been looking at the 2N voiceblue products. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP GSM Gateways
I’m looking at setting up a VoIP GSM gateway to connect to my asterisk box. What experience have people on this list have with GSM gateway hardware. I have been looking at the 2N voiceblue products. We are using the voiceblue that supports a maximum of 4 x sims (and are using all four sims), for the pass few months without issue with different versions of Asterisk, including the current version (1.2.13), which has allowed to take full advantage of our corporate mobile account . Our desk phones are SNOM 320's and 360's. But the Asterisk server is the media gateway path for all communications. smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] VoIP GSM Gateways
Peter, How much does the 4 port cost? How many simultaneous calls can you make? Do you need a mobile account from a mobile provider such as T-mobile? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter J Dean Sent: Saturday, 28 October 2006 7:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] VoIP GSM Gateways I'm looking at setting up a VoIP GSM gateway to connect to my asterisk box. What experience have people on this list have with GSM gateway hardware. I have been looking at the 2N voiceblue products. We are using the voiceblue that supports a maximum of 4 x sims (and are using all four sims), for the pass few months without issue with different versions of Asterisk, including the current version (1.2.13), which has allowed to take full advantage of our corporate mobile account . Our desk phones are SNOM 320's and 360's. But the Asterisk server is the media gateway path for all communications. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to make different ext using different trunks?
Hi, I want to do so that extension 501 will always use trunk1 for outbound calls and 502 will use trunk2 for outboud calls. How do I do this. Right now all extensions use the same trunk for outbound calls. -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
