Re: [asterisk-users] Asterisk room monitor
Thanks.
On 4/13/2010 3:07 AM, Ioan Indreias wrote:
On Mon, Apr 12, 2010 at 8:19 PM, Mark Hulberasterisk.ad...@hulber.com
wrote:
I want to use a voip speaker phone as a room monitor. Requirements:
A phone that I can set to auto answer in speaker mode.
A phone with a good speaker phone.
Ability to make the audio one way. I want to monitor the room but not
have my voice heard in the room. Yes, the mute button can accomplish
this also.
I have been using the SPA942's around the house (the speaker is just ok
but probably good enough). Can I set one of these or a similar Cisco
phone to auto answer in speaker mode? Any ideas on an alternative phone
that would allow this?
The alternative is to just set up the call locally and then leave the
room with the line open but ideally I'd like to be able to open up the
monitor on demand.
Thanks,
MARK.
Hello Mark,
Please find bellow a dialplan proof-of-concept for your requirement
(is based on intercom module present in FreePBX and adapted to have
only one way audio for 60 secconds). We have tested with Linksys
SPA9XX phones and works fine (hint: clear regional=call progres
tones=page tone in order to cancel the page tone if you need to be
super-silent).
HTH,
Ioan Indreias
www.modulo.ro
exten = _6XX,1,Answer
exten = _6XX,n,Set(_ALERTINFO=Alert-Info: Ring Answer)
exten = _6XX,n,Set(_CALLINFO=Call-Info:uri\;answer-after=0)
exten = _6XX,n,Set(_SIP_URI_OPTIONS=intercom=true)
exten = _6xx,n,SipAddHeader,${ALERTINFO})
exten = _6XX,n,SipAddHeader,${CALLINFO})
exten = _6XX,n,Dial(SIP/1${EXTEN:1},5,G(100))
exten = _6XX,100,Goto(200)
exten = _6XX,101,Goto(300)
exten = _6XX,200,ChanSpy(SIP/1${EXTEN:1})
exten = _6XX,300,Wait(60)
exten = _6XX,n,Hangup
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[asterisk-users] Asterisk room monitor
I want to use a voip speaker phone as a room monitor. Requirements: A phone that I can set to auto answer in speaker mode. A phone with a good speaker phone. Ability to make the audio one way. I want to monitor the room but not have my voice heard in the room. Yes, the mute button can accomplish this also. I have been using the SPA942's around the house (the speaker is just ok but probably good enough). Can I set one of these or a similar Cisco phone to auto answer in speaker mode? Any ideas on an alternative phone that would allow this? The alternative is to just set up the call locally and then leave the room with the line open but ideally I'd like to be able to open up the monitor on demand. Thanks, MARK. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Losing local SIP phones when internet goes down?
I have the same problem. I have asterisk on the public internet and other ips on the private lan. When the internet goes down my private asterisk network is compromised. My thought is that it has something to do with the ports/ips on which asterisk is trying to communicate. It may be a configuration issue but as of yet I haven't figured it out. On 2/4/2010 9:05 PM, Nikhil Nair wrote: Hi, I'm getting some strange behaviour on Asterisk 1.4 running on Debian Stable (Lenny). I suspect it's something to do with my setup, rather than a bug, but I'm struggling to see it, and would appreciate any input. Setup: PC with two ethernet cards: eth0 goes to local network, including two SIP phones (Aastra 9112i, wired, and Nokia E75, over WIFI); eth1 goes to router and thence to the internet over ADSL. PC also has one Zap channel. the SIP phones use DHCP but have defined IP addresses (DHCP server running on the PC). The PC is also running a firewall (FIAIF), but not a DNS server. Version of Debian Asterisk package: 1:1.4.21.2~dfsg-3+lenny1 Problem: When the internet connection goes down (which has been happening sporadically of late), connections to the two SIP phones on the local network get lost; ongoing calls from one of these phones over the Zap channel may get terminated, despite not using the internet. I can reproduce this by switching off my ADSL router; however, if I simply take down the eth1 interface completely (by using ifdown eth1, which executes route del default gw ... eth1 and ifconfig eth1 down), the connections to the two SIP phones continue with no problems at all. I enclose an extract from my sip.conf below. Also, the logs indicate that Asterisk thinks the SIP phones are no longer reachable (ping timing out), while a manual ping from the same machine shows no trouble at all: the wired phone is responding in less than 2 ms each time, while the wireless one was a max of about 120 ms. Any thoughts much appreciated! Hopefully it's something obvious that I've overlooked... Oh, BTW, the local phones are on a private net (10.9.8.xxx), but as it's the Asterisk box that's doing the NAT'ing, I used nat=no; I presume that's correct. eth0 has address 10.9.8.1, while eth1 has a global internet IP address. Cheers, Nikhil. - Extract from sip.conf: [general] context=incoming srvlookup=yes realm=nikhil-nair.net ; Various register= statements, not relevant to the local phones [101] ; Aastra 9112i at 10.9.8.101 type=friend secret=... qualify=yes ; Qualify peer is no more than 2000 ms away nat=no ; This phone is not natted host=dynamic ; This device registers with us canreinvite=no ; Asterisk by default tries to redirect context=local disallow=all allow=alaw allow=ulaw [111] ; Nokia E75 via WIFI access point, at 10.9.8.111 type=friend secret=... qualify=yes ; Qualify peer is no more than 2000 ms away nat=no ; This phone is not natted host=dynamic ; This device registers with us canreinvite=no ; Asterisk by default tries to redirect context=local disallow=all allow=alaw allow=ulaw allow=gsm -- _ -- 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] disable comfort noise
This is how I understand it. The other end is trying to set up comfort noise and asterisk is letting you know that it's trying to do so and maybe you can turn this off on the other end. I have a particular voip provider where I get this message. I think if you get it turned off there's a little bit better performance on the connection. By now people are getting used to calls without comfort noise but for a long time it threw people off because they weren't sure if the call was still connected. On 1/30/2010 1:30 AM, uzzi wrote: On Fri, Jan 29, 2010 at 1:14 PM, ad...@3a.hu mailto:ad...@3a.hu wrote: To get back to the original poster's possible situation, i've seen this with my first IP phone, which was a cisco 7912 (SIP image). With that phone, asterisk sometimes gave me this same error. I'm quite sure i've asked the very same question here back then (probably i was a bit more specific :). Since it is related to only this type of phone, i've gone to different ip phone products. regards adam -- _ -- 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 Please correct me if I'm wrong As the error says, Please turn off on client if possible. Comfort noise (aka silent suppression, or Voice Activity Detection (VAD)) is not supported by Asterisk. It needs to be turned off on the user (client) end. This may be a phone or another switch/PBX. See http://www.voip-info.org/wiki/view/RTP+Silence+Suppression for more details -- _ -- 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] ASTSBINDIR not being picked up by safe_asterisk
When I run make install I don't see this file getting overwritten. Do I have to delete it to get this to happen? On 1/25/2010 7:06 PM, Tilghman Lesher wrote: On Monday 25 January 2010 08:52:45 Mark Hulber wrote: Recently safe_asterisk is failing to pick up ASTSBINDIR. I've never had this problem before and even when I move to back versions I have the issue. I did upgrade safe_asterisk and the init.d scripts a version or so ago but even when I try older ones I still have the problem. When I hard code the location things seem to work. The problem that occurs is: cat: __ASTERISK_VARRUN_DIR__/asterisk.pid: No such file or directory Automatically restarting Asterisk. But I think this is just a side effect of not finding asterisk in the /usr/sbin directory in the first place. Anyone run across this or have an idea what might have happened? I don't know if it was a Redhat update issue or some change in my configuration or what. When I make the following change in safe_asterisk it works ok: ASTSBINDIR=__ASTERISK_SBIN_DIR__ ASTSBINDIR=/usr/sbin Sounds like you manually copied the safe_asterisk script to /usr/sbin, instead of relying on 'make install' to do it for you. The install target does some extra processing of the script for you. -- _ -- 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] ASTSBINDIR not being picked up by safe_asterisk
Great, do you know of any other files outside of /usr/lib/asterisk/modules that get recreated? I also place rc.redhat.asterisk as asterisk in /etc/rc.d/init.d I don't see that safe_asterisk_restart gets placed anywhere. It looks like astgenkey and autosupport both get written over. On 1/26/2010 11:15 AM, Tilghman Lesher wrote: On Tuesday 26 January 2010 10:08:39 Mark Hulber wrote: On 1/25/2010 7:06 PM, Tilghman Lesher wrote: On Monday 25 January 2010 08:52:45 Mark Hulber wrote: Recently safe_asterisk is failing to pick up ASTSBINDIR. I've never had this problem before and even when I move to back versions I have the issue. I did upgrade safe_asterisk and the init.d scripts a version or so ago but even when I try older ones I still have the problem. When I hard code the location things seem to work. The problem that occurs is: cat: __ASTERISK_VARRUN_DIR__/asterisk.pid: No such file or directory Automatically restarting Asterisk. But I think this is just a side effect of not finding asterisk in the /usr/sbin directory in the first place. Anyone run across this or have an idea what might have happened? I don't know if it was a Redhat update issue or some change in my configuration or what. When I make the following change in safe_asterisk it works ok: ASTSBINDIR=__ASTERISK_SBIN_DIR__ ASTSBINDIR=/usr/sbin Sounds like you manually copied the safe_asterisk script to /usr/sbin, instead of relying on 'make install' to do it for you. The install target does some extra processing of the script for you. When I run make install I don't see this file getting overwritten. Do I have to delete it to get this to happen? Correct. It's only created if it doesn't already exist. -- _ -- 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] ASTSBINDIR not being picked up by safe_asterisk
Recently safe_asterisk is failing to pick up ASTSBINDIR. I've never had this problem before and even when I move to back versions I have the issue. I did upgrade safe_asterisk and the init.d scripts a version or so ago but even when I try older ones I still have the problem. When I hard code the location things seem to work. The problem that occurs is: cat: __ASTERISK_VARRUN_DIR__/asterisk.pid: No such file or directory Automatically restarting Asterisk. But I think this is just a side effect of not finding asterisk in the /usr/sbin directory in the first place. Anyone run across this or have an idea what might have happened? I don't know if it was a Redhat update issue or some change in my configuration or what. When I make the following change in safe_asterisk it works ok: ASTSBINDIR=__ASTERISK_SBIN_DIR__ ASTSBINDIR=/usr/sbin Here are my version levels: Asterisk 1.6.2.1 built by root on a x86_64 running Linux on 2010-01-15 16:22:39 UTC Linux 2.6.18-164.11.1.el5 #1 SMP Wed Jan 6 13:26:04 EST 2010 x86_64 x86_64 x86_64 GNU/Linux MARK. -- _ -- 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] Extra Sounds Missing on 1.6.1.6 install
It looks like there's a problem with the location or naming of the Extra SLN16 sounds: --14:11:43-- http://downloads.digium.com/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz Resolving downloads.digium.com... 76.164.171.232 Connecting to downloads.digium.com|76.164.171.232|:80... connected. HTTP request sent, awaiting response... 301 Moved Permanently Location: http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz [following] --14:11:44-- http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz Resolving downloads.asterisk.org... 76.164.171.233 Connecting to downloads.asterisk.org|76.164.171.233|:80... connected. HTTP request sent, awaiting response... 404 Not Found 14:11:44 ERROR 404: Not Found. make[1]: *** [/var/lib/asterisk/sounds/.asterisk-extra-sounds-fr-SLN16-1.4.9] Error 1 make[1]: Leaving directory `/usr/src/asterisk-1.6.1.6/sounds' make: *** [datafiles] Error 2 [r...@asterisk asterisk]# make menuselect make[1]: Entering directory `/usr/src/asterisk-1.6.1.6' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extra Sounds Missing on 1.6.1.6 install
Looks like the Makefile is broken and putting SLN16 instead of sln16. Mark Hulber wrote: It looks like there's a problem with the location or naming of the Extra SLN16 sounds: --14:11:43-- http://downloads.digium.com/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz Resolving downloads.digium.com... 76.164.171.232 Connecting to downloads.digium.com|76.164.171.232|:80... connected. HTTP request sent, awaiting response... 301 Moved Permanently Location: http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz [following] --14:11:44-- http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-extra-sounds-fr-SLN16-1.4.9.tar.gz Resolving downloads.asterisk.org... 76.164.171.233 Connecting to downloads.asterisk.org|76.164.171.233|:80... connected. HTTP request sent, awaiting response... 404 Not Found 14:11:44 ERROR 404: Not Found. make[1]: *** [/var/lib/asterisk/sounds/.asterisk-extra-sounds-fr-SLN16-1.4.9] Error 1 make[1]: Leaving directory `/usr/src/asterisk-1.6.1.6/sounds' make: *** [datafiles] Error 2 [r...@asterisk asterisk]# make menuselect make[1]: Entering directory `/usr/src/asterisk-1.6.1.6' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID providers in Toronto
I've had a good ongoing experience using http://www.unlimitel.ca. They are responsive and reliable. MARK. Asterisk guy wrote: hi Can anyone recommend a good DID provider offering numbers in Toronto ? ( 1 very stable 2 support porting numbers from Bell, primus, telus.. ) Mario ___ --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] Sip phones using the wrong context for an outbound call
It might help to show your Support context in outbound.conf. MARK. Alexander Topolanek wrote: Hi, recently I changend a few things in the configuration of the Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-d of a customer. One demand was that different groups of SIP-Phones are using different trunks to the outside worls, so I moved some of them to a Support context. However, dial out from this phones failes as they're still looking for an extension in the default context, which doesn't exist ([default] is now pretty crippled for security reasons). Is there a way to see in which context a peer (wether SIP or Zap or whatever) starts? This is how the configuration for the extension typically looks: ; Grandstream [61] type=peer username=61 secret=xxx context=Support reinvite=no canreinvite=no host=dynamic subscribecontext=Support ;[EMAIL PROTECTED] ;allow=alaw ;allow=ulaw ;allow=g723.1 best regards ___ --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] Sending '#' with Dial
Have you tried setting the CALLERID variables? If the provider is ignoring those then I guess they are asking you to set per call blocking? I don't know how to do that. exten = s,1,Set(CALLERID(number)=3025551212|a) exten = s,n,Set(CALLERID(name)=Joe Smith|a) MARK. Emil Thelin wrote: Hi! I have a working asterisk-setup with four sip-clients. Everything works great but when the users call someone the phonenumber shows up on the receiving ends callerid-display. To correct this my provider told me to send #31# before the phonenumber, tried this with: Dial(SIP/[EMAIL PROTECTED]) but my asterisk tells me that it isn't a valid extension. The INVITE looks fine, '#31#phonenumber@provider' but my provider then sends SIP/2.0 404 Not Found back to me. Any thoughts? /e -- http://hostname.nu/~emil ___ --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] MWI not working in 1.4
Before I open a bug I'll ask again if anyone else is having trouble with receiving MWI on SIP devices in 1.4. My configuration was working fine in 1.2 but as soon as I change to any build of 1.4 I don't get notification on any of several SIP devices. I can post my configuration but since it was working I can only assume it would break if something in voicemail.conf has changed or sip.conf but current examples appear to concur with my setup. From my peer definition: [EMAIL PROTECTED],[EMAIL PROTECTED],[EMAIL PROTECTED] subscribemwi=yes voicemail.conf [mainmenu] 100 = 1234,User 1,[EMAIL PROTECTED],,saycid=no|envelope=no|review=yes|tz=eastern 200 = 1234,User 2,[EMAIL PROTECTED],,saycid=yes|envelope=yes|review=yes|tz=eastern 300 = 1234,User 3,[EMAIL PROTECTED],,saycid=yes|envelope=yes|review=yes|tz=eastern ___ --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] 480i phone: Is there a trick to registering with *??
I set up mine with the web interface but I notice that some settings can only be made by config files. Do you know how to extract the current config file from the phone? Here's how I set up the web interface: Authentication Name: aastra480_1 Password: password BLA Number: blank Line Mode: Generic Proxy Server: 192.168.0.80 Proxy Port: 5060 Outbound Proxy Server: 192.168.0.80 Outbound Proxy Port: 5060 Registrar Server: 192.168.0.80 Registrar Port: 5060 Registration Period: 300 Dave Cotton wrote: On Sat, 2006-09-30 at 09:35 +0200, Dave Cotton wrote: and 00085D183552.cfg (not uppercase) contains Whoops note ___ --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] MWI on 1.4 Beta
Anyone else having trouble with MWI on 1.4 Beta? The messages are getting stored and I'm getting the emails but no stutter tone or MWI as far as I can tell. MARK. ___ --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] Building Zaptel 1.2.9 with Octasic
Any pointers about on how to get around this build problem in Zaptel 1.2.9? /usr/src/zaptel-1.2.9/wct4xxp/fw2h /usr/src/zaptel-1.2.9/wct4xxp/OCT6114-128D.ima /usr/src/zaptel-1.2.9/wct4xxp/vpm450m_fw.h make[3]: *** No rule to make target `/usr/src/zaptel-1.2.9/wct4xxp/../oct612x/include/oct6100api/oct6100_api.h', needed by `/usr/src/zaptel-1.2.9/wct4xxp/vpm450m.o'. Stop. make[2]: *** [/usr/src/zaptel-1.2.9/wct4xxp] Error 2 make[1]: *** [_module_/usr/src/zaptel-1.2.9] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.EL-i686' make: *** [linux26] Error 2 ___ --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] Building Zaptel 1.2.9 with Octasic
Yes, it worked. I didn't get the announcement of 1.2.9.1. MARK. Tzafrir Cohen wrote: On Wed, Sep 13, 2006 at 12:00:27PM -0400, Mark Hulber wrote: Any pointers about on how to get around this build problem in Zaptel 1.2.9? Get 1.2.9.1, that has fixed exactly that. (and improvd Astribank drivers, thanks Kevin) ___ --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] Set DID?
Hey Dean,
Maybe it would be easier if you would describe what you would like to
happen as a result of doing what you are asking. When you have an
incoming call from this provider do you know what number was dialed? Are
you expecting this number to be displayed somewhere or are you looking
to take an action based on it?
MARK.
Dean Collins wrote:
Hi Mouta, sorry…can you elaborate a little (maybe something a little
more basic).
Cheers,
Dean
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Marco
Mouta
*Sent:* Thursday, 10 August 2006 2:16 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Set DID?
Handle incoming calls to s extension and in the next priority set
EXTEN var to your DID then make a goto to desired context.
Hope it helps,
MoutaPT
On 8/10/06, *Dean Collins* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi Eric,
No I know what I want. I want to set the DID to be 212-531-6214 as my
current provider doesn't send a DID number.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
[mailto:asterisk-users- mailto:asterisk-users-
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf
Of Eric ManxPower Wieling
Sent: Thursday, 10 August 2006 1:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set DID?
Dean Collins wrote:
Is there a command for setting of a DID number?
Eg below I can set callerid
[custom-fromiaxfwd]
exten = s,1,Set(CALLERID(number)=2125316214)
Butw what I would prefer to do is set DID -like this (it doesn't
work
[custom-fromiaxfwd]
exten = s,1,Set(CALLERDID(number)=2125316214)
I couldn't find anything in the voip-info commands section so was
hoping
for a clue from the list.
You are trying to set the CallerID, not setting the DID. The DID is
in
${EXTEN}.
If you want to set the CallerID for calls to the PSTN you must be
using
ISDN. If you are using VoIP, then the VoIP server must be using ISDN
(pretty much all of them are).
Your carrier must permit you to set that info. Not all of them do.
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and Montgomery.
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Com os melhores cumprimentos,
Marco Mouta
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Re: [asterisk-users] DTMF-CallerID on POTS
I was using zap but I ditched the PSTN for now. Try taking a look at:
CALLERID(name) or CALLERID(number) instead.
MARK.
Greg Delgado wrote:
Has anyone got a working analog connection to POTS
wherein DTMF, *not* FSK is used to send caller id by
the telco switch towards asterisk?
I've tried Asterisk 1.2.10, SVN trunk, and SVN branch
but so far has been unsuccesful.
When asterisk receives a call, I can see from DEBUG
that chan_zap is able to pick up all DTMF digits like
so:
Aug 11 17:06:17 DEBUG[5113]: chan_zap.c:4624 zt_read:
DTMF digit: D on Zap/1-1
Aug 11 17:06:17 DEBUG[5113]: chan_zap.c:4624 zt_read:
DTMF digit: 0 on Zap/1-1
Aug 11 17:06:17 DEBUG[5113]: chan_zap.c:4624 zt_read:
DTMF digit: 2 on Zap/1-1
Aug 11 17:06:17 DEBUG[5113]: chan_zap.c:4624 zt_read:
DTMF digit: 8 on Zap/1-1
Aug 11 17:06:18 DEBUG[5113]: chan_zap.c:4624 zt_read:
DTMF digit: 1 on Zap/1-1
Aug 11 17:06:18 DEBUG[5113]: chan_zap.c:4624 zt_read:
DTMF digit: 1 on Zap/1-1
Aug 11 17:06:18 DEBUG[5113]: chan_zap.c:4624 zt_read:
DTMF digit: 8 on Zap/1-1
Aug 11 17:06:18 DEBUG[5113]: chan_zap.c:4624 zt_read:
DTMF digit: 8 on Zap/1-1
Aug 11 17:06:18 DEBUG[5113]: chan_zap.c:4624 zt_read:
DTMF digit: 9 on Zap/1-1
Aug 11 17:06:18 DEBUG[5113]: chan_zap.c:4624 zt_read:
DTMF digit: C on Zap/1-1
but the number somehow does not get passed to
${CALLERID} variable.
in zaptel.conf:
callerid=asreceived
cidsignalling=dtmf
cidstart=polarity
does someone have a similar system that's working? I'd
like to know which asterisk and zaptel versions you
are running
Thanks,
Greg
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[Asterisk-Users] Voipjet Problem?
I started to have a problem today that all my calls through voipjet result in just timing out after my assigned timeout period. I tried multiple of their servers with the same problem. Anyone else having a problem? I am running: Asterisk SVN-branch-1.2-r24381M built by root @ asterisk.hulber.com on a i686 running Linux on 2006-05-03 14:14:07 UTC I can connect with other IAX providers. MARK. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] really need help with outgoing calls..PSTN errors
Have you tried dialing an 800 number? Does that work? This extension:
exten = _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
seems to be missing one X since it's only 10 digits long. Your PSTN
probably requires a 1 to be dialed also. On the other hand,
exten = _91[1234567]XXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
you should probably be matching this extension instead although you
won't be able to match anywhere that has an area code that starts with
an 8 or 9. (905, 916, 914 as a few examples).
MARK.
sdgesa gaeharth wrote:
I cant seem to get outgoing calls to be placed properly .. No matter
what I try I get an error from the PSTN company saying that the call
can not be completed as dialed or you need to dial a one... The
asterisk debugging seems to show the correct number being dialed out
of the zap interface... the 9 is being stripped and there is a 1
where it is supposed to be. I am thinking it is a problem between the
zap interface and the PSTN.
thanks
extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
ATTENDANT=1001
OUTBOUNDTRUNK=ZAP/g1
[extentions]
exten = _10XX,1,Ringing
exten = _10XX,2,Dial(SIP/${EXTEN},20)
exten = _10XX,3,Answer
exten = _10XX,4,VoiceMail([EMAIL PROTECTED]
mailto:[EMAIL PROTECTED])
exten = _10XX,5,Hangup
[voicemail]
exten = _910XX,1,Wait(1)
exten = _910XX,2,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])
[local]
include = extentions
include = voicemail
[incoming]
exten = s,1,Answer
exten = s,n,Wait(2)
exten = s,n,Set(TIMEOUT(response)=15)
exten = s,n,Background(company-intro)
exten = s,n,WaitExten()
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup()
exten = 0,1,Dial(SIP/${ATTENDANT},20)
exten = 1,1,Directory(voicemail,extentions,f)
exten = 2,1,Directory(voicemail,extentions)
exten = 1234,1,Playback(abandon-all-hope)
include = extentions
exten = i,1,Playback(vm-goodbye)
exten = i,2,Hangup()
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup()
[outbound]
ignorepat = 9
exten = _9XX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _9XX,2,Congestion()
exten = _9XX,102,Congestion()
exten = _91800NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91800NXX,2,Congestion()
exten = _91800NXX,102,Congestion()
exten = _91888NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91888NXX,2,Congestion()
exten = _91888NXX,102,Congestion()
exten = _91877NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91877NXX,2,Congestion()
exten = _91877NXX,102,Congestion()
exten = _91866NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91866NXX,2,Congestion()
exten = _91866NXX,102,Congestion()
exten = _91900NXX,1,Congestion()
exten = _91976NXX,1,Congestion()
exten = _91[1234567]XXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91[1234567]XXNXX,2,Congestion()
exten = _91[1234567]XXNXX,102,Congestion()
exten = 9911,1,Dial(${OUTBOUNDTRUNK}/911)
exten = 9411,1,Dial(${OUTBOUNDTRUNK}/411)
exten = 0,1,Dial(${OUTBOUNDTRUNK}/0)
[local-access]
include = local
include = outbound
zapata.conf:
[channels]
group = 1
language=en
context=incoming
signalling=fxs_ks
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
callerid = Dulles Micro, LLC 703 450 5000
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
channel = 1
zaptel.conf:
fxsks=1,2,3,4
loadzone = us
defaultzone=us
Brings words and photos together (easily) with
PhotoMail
http://us.rd.yahoo.com/mail_us/taglines/PMall/*http://photomail.mail.yahoo.com
- it's free and works with Yahoo! Mail.
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Re: [Asterisk-Users] TIMESTAMP, DATETIME not working
show function STRFTIME
-= Info about function 'STRFTIME' =-
[Syntax]
STRFTIME([epoch][,[timezone][,format]])
[Synopsis]
Returns the current date/time in a specified format.
[Description]
Not available
Example:
exten = s,n,NoOp(TIME=${STRFTIME(,EST5EDT,%d%b%Y-%H:%M:%S)})
MARK.
Corporate IT Solutions - Michael Dunne wrote:
I am using the latest SVN version 1.2 of Asterisk
When I attempt to test the output of certain variables, for use in file
naming etc, certain key ones appear to fail.
exten = ,1,NoOp(${EPOCH})
Returns
-- Executing NoOp(SIP/200-638c, 1141352935) in new stack
exten = 5556,1,NoOp(${TIMESTAMP})
Returns
-- Executing NoOp(SIP/200-8cc9, ) in new stack
exten = 5557,1,NoOp(${DATETIME})
Returns
-- Executing NoOp(SIP/200-83ca, ) in new stack
Epoch works fine, however none of the other human readable timestamps
seem to be working. Is there anything else required to initialise them,
or how can I test to make sure they are being initialised correctly.
Thanks
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Re: [Asterisk-Users] FW: Re: Delay on Phone ringing
The only time I see recorded in your log is that of the recording check -- Executing AGI(Zap/1-1, recordingcheck|20060227-131600|1141046151.2) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060227-131600|1141046151.2: Inbound recording not enabled which doesn't seem to take any time. Only you would know at what phase the dialplan was in at each point of the 12 seconds. How long did it take before this took place: -- Starting simple switch on 'Zap/1-1' How long did this phase take: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- AGI Script dialparties.agi completed, returning 0 MARK. Ash Thakrar wrote: Hi, I have just joined this mail list yesterday and have been searching the Asterisk wiki prior to posting this question. Unfortunately I am not sure if I am searching at the correct places, so I do apologise if this has been posted before. I have currently been tasked to roll out VoIP phones through out our office as the current proprietary Panasonic PBX has no more channels. Thus I have installed [EMAIL PROTECTED] on VIA SP13000,512Mb Ram and using 2 x Digum TDM400P cards with both having 4x TDM40B FXO modules. I have rolled out 12 x Snom320 phones 1 x Snom360 in the office. For the test phase, I wanted to use the current PBX, Therefore Port 1 of the TDM is currently connected to one of the POTS extensions which is spare on the current PBX. Current problems I am facing in the test phase: Whenever I call from outside e.g. from the fax line (separate line) or my mobile, to the main number setup on the Trunk, I get a delay of around 12sec before the VoIP phone actually rings, although the phones connected to the current PBX, ring immediately. I have attached the output file and noticed that the DBget is trying to find ‘something’ in the AstDB, would that be causing the delay? Or am I looking at the wrong place altogether. Please Help Regards Ash Thakrar asterisk1*CLI soft hangup Zap/1-1 Requested Hangup on channel 'Zap/1-1' == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Zap/1-1' in macro 'exten-vm' == Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' -- Executing GotoIf(Zap/1-1, 1?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf(Zap/1-1, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(Zap/1-1, ) in new stack -- Executing Wait(Zap/1-1, 1) in new stack -- Executing SetVar(Zap/1-1, intype=EXT-220) in new stack -- Executing Cut(Zap/1-1, intype=intype|-|1) in new stack -- Executing GotoIf(Zap/1-1, 1?7:9) in new stack -- Goto (from-pstn-reghours,s,7) -- Executing Wait(Zap/1-1, 3) in new stack -- Executing Goto(Zap/1-1, ext-local|220|1) in new stack -- Goto (ext-local,220,1) -- Executing Macro(Zap/1-1, exten-vm|novm|220) in new stack -- Executing Macro(Zap/1-1, user-callerid) in new stack -- Executing DBget(Zap/1-1, AMPUSER=DEVICE//user) in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=/user -- DBget: Value not found in database. -- Executing DBget(Zap/1-1, AMPUSERCIDNAME=AMPUSER//cidname) in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname -- DBget: Value not found in database. -- Executing GotoIf(Zap/1-1, 1?5) in new stack -- Goto (macro-user-callerid,s,5) -- Executing NoOp(Zap/1-1, Using CallerID ) in new stack -- Executing SetVar(Zap/1-1, FROMCONTEXT=exten-vm) in new stack -- Executing Macro(Zap/1-1, record-enable|220|IN) in new stack -- Executing GotoIf(Zap/1-1, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(Zap/1-1, recordingcheck|20060227-131600|1141046151.2) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060227-131600|1141046151.2: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(Zap/1-1, No recording needed) in new stack -- Executing Macro(Zap/1-1, dial|15|tr|220) in new stack -- Executing GotoIf(Zap/1-1, 0?4:2) in new stack -- Goto (macro-dial,s,2) -- Executing GotoIf(Zap/1-1, 0?5:4) in new stack -- Goto (macro-dial,s,4) -- Executing AGI(Zap/1-1, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- dialparties.agi: priority = 4 -- dialparties.agi: callingani2 = 0 -- dialparties.agi: accountcode = -- dialparties.agi: channel = Zap/1-1 -- dialparties.agi: callerid = unknown -- dialparties.agi: context = macro-dial -- dialparties.agi: callington = 0 -- dialparties.agi: dnid = unknown -- dialparties.agi:
Re: [Asterisk-Users] Re: delaying answer for a number of rings or an amount
It sounds like you both need a Zap card. You can ring the analog phone and/or the Sip phones when a call comes in on the POTS line that is connected to the card. MARK. Brian J. Murrell wrote: On Fri, 2006-02-03 at 07:37 -0700, Bromont Quebec wrote: Well in my setup I have a few IP phones connected to Asterisk as well as POTS phones on my analog line. Ahhh. So we share the latter at least. When a call for my daughter comes in on the analog line (determined from callerID) I send it to her own voicemail after 20 seconds of ringing. It all works quite well. Hrm. Yeah, this is what I'm trying to do. Here's a step-by-step of what happens below: 1 - a call comes in and Asterisk rings SIP/Brian and SIP/joe for 30 seconds. So you don't want Asterisk to wait and see if the POTS line is picked up before ringing the SIP phones? Interesting. 2 - After 30 seconds if the line is still ringing (nobody picked up POTS phone or SIP phones) * answers the line and sends to Voicemail. Asterisk never picks up the call until the 30 seconds are up. What seems to be happening here is that even if somebody picks up the POTS line within a few seconds, after the 30 seconds (Wait() in my case, but I'd imagine the same will happen after ringing the SIP lines for 30s) is up Asterisk is also on the POTS line (with the callee who picked up the POTS phone) doing the voicemail intro and recording the conversation. [from-pots] exten = s,1,Dial(SIP/brianSIP/joe,30) exten = s,2,Voicemail(u2001) exten = s,3,Hangup I will try this exactly and see if it works any better. b. ___ --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] DTMF Sporadicaly Being Generated
I've been having horrible DTMF problems lately on from Sipura ATAs to ZAP and IAX. It's primarily with repeated digits. I'm starting to move my connections to SIP until I can get it all figured out. Other than updating to the newest SVN trunk I haven't made changes on my end that should have caused this. I've already put some of my IAX debug on a bug report relating to double dtmf with Jitterbuffer enabled. MARK. Kevin P. Fleming wrote: Michael L. Young wrote: I have a TE411P card in my * box. I am running FC4 x86_64. I used to have two TE110 cards in the same box that worked without any problems. Since changing to the TE411P cards, I am getting random DTMF tones being produced on a bridged connection through the same Channel Bank that I was using before upgrading to the TE411P. This is a known problem, been discussed on the lists many times. You should contact Digium Support, since you just purchased a Digium card. They are best equipped to handle issues related to Digium hardware. ___ --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] DTMF Sporadicaly Being Generated
Good to know. I was able to play around and get it mostly working but I'm still not able to get DTMF working with Jitterbuffer ON for IAX although I previously could at least with some providers. I had to define my SIP extensions to use INBAND and set the Sipura devices to also use INBAND and not process INFO or AVT. I also noticed I was using dtmf= instead of dtmfmode= which may or may not be in my imagination that the latter works better (or at all). Now at least people can listen to voicemail and authenticate a remote conference call using the same device. MARK. Rob Thomas wrote: To quote Kevin: DTMF handling in the trunk is in a state of flux right now. It won't be resolved until this weekend. Don't use SVN for a production system, it's lots broken right now. If you really must, stick with r8786 for a while. --Rob -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mark Hulber Sent: Sunday, 5 February 2006 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DTMF Sporadicaly Being Generated I've been having horrible DTMF problems lately on from Sipura ATAs to ZAP and IAX. It's primarily with repeated digits. I'm starting to move my connections to SIP until I can get it all figured out. Other than updating to the newest SVN trunk I haven't made changes on my end that should have caused this. I've already put some of my IAX debug on a bug report relating to double dtmf with Jitterbuffer enabled. MARK. Kevin P. Fleming wrote: Michael L. Young wrote: I have a TE411P card in my * box. I am running FC4 x86_64. I used to have two TE110 cards in the same box that worked without any problems. Since changing to the TE411P cards, I am getting random DTMF tones being produced on a bridged connection through the same Channel Bank that I was using before upgrading to the TE411P. This is a known problem, been discussed on the lists many times. You should contact Digium Support, since you just purchased a Digium card. They are best equipped to handle issues related to Digium hardware. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] instant fallback to zap in case of sip/iax/xyz-failure
My experience is that when an iax or sip channel is unavailable for some
reason it fails right away despite whatever timeout I have set for the
call. In these cases the caller doesn't even realize that the call has
failed over to the next carrier.
exten = s,n(dial1),Dial(${VOIPJET}/${ARG1}|90,T)
exten = s,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?dial2)
exten = s,n,Macro(rhangup)
exten = s,dial1+101,GotoIf($[${DIALSTATUS} = BUSY]?s-BUSY|1)
exten = s,n(dial2),Dial(IAX2/[EMAIL PROTECTED]/${ARG1}|90,T)
exten = s,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?dial3)
MARK.
Christian Benke wrote:
i would like to carry some oversea pstn-destinations via sip to providers
like stanaphone, however, in case of a network-failure or if the provider
is not available, i want to fallback to the zap-channels so the call is
carried out to the pstn directly.
the usual approach would be to check the dialstatus(e.g.NOANSWER).
however, asterisk tries 60seconds to reach that peer(even when the ip i'm
sending the call too is a dead end(no host)). i could limit a call by
setting a timeout but this limit would also apply if a final destination
doesn't pick up within the timeout.
so basically, when i send a call via a sip-channel, i would like to know
the network-status of the foreign host immediately(at least within 5
seconds) so i can reroute the call without having to wait for a host that
is probably dead...
this seems to be possible with iax and CHANUNAVAIL,
(http://bugs.digium.com/bug_view_advanced_page.php?bug_id=3360history=1),
though i haven't tried it.
also i _need_ to use sip, iax (currently) is not an option.
is there any mechanism in asterisk that allows to get the vital sip-status
of a foreign host?! thanks for your input!!! ;-)
regards
christian
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[Asterisk-Users] CALLERIDNAME/CALLERIDNUM Deprecation
Previously, when I wanted to forward to incoming callerid when I
forwarded a call to another number I had to set the callerid on the
outgoing call to be that of the incoming number. So today I do this:
exten = s,n,Set(CALLERID(name)=${CALLERIDNAME})
because I want the outgoing callerid that I forward to not be the normal
callerid of the local extension but I want to forward the incoming
callerid. Now that CALLERIDNAME is deprecated, how do I differentiate
between the CALLERID on the incoming channel and the callerid set on the
outgoing channel? The deprecation advice seems to suggest that I change
my set statement to:
exten = s,n,Set(CALLERID(name)=CALLERID(name))
which doesn't clearly make any sense to me. The function info suggests
there is an optional-CID parameter but I don't know what the options are.
MARK.
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Re: [Asterisk-Users] CALLERIDNAME/CALLERIDNUM Deprecation
I have an incoming call on one channel coming into asterisk, and I'm
forwarding the call using Dial on several other channels such as to
reach a cell phone and work. I don't want the caller ID that has been
assigned to the outbound SIP or IAX account to show up on the cell phone
but the callerid of the original incoming caller.
MARK.
Kevin P. Fleming wrote:
Mark Hulber wrote:
exten = s,n,Set(CALLERID(name)=${CALLERIDNAME})
This could never have accomplished anything, since those two
references affect the exact same variable internally.
because I want the outgoing callerid that I forward to not be the
normal callerid of the local extension but I want to forward the
incoming callerid. Now that CALLERIDNAME is deprecated, how do I
differentiate between the CALLERID on the incoming channel and the
callerid set on the outgoing channel? The deprecation advice seems
to suggest that I change my set statement to:
exten = s,n,Set(CALLERID(name)=CALLERID(name))
You'll have to more clearly define what you want to accomplish;
normally, the Dial() application sets the CLID/CNAM info on the
outgoing channel based on what is present on the channel placing the
call.
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Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
The paper is definitely interesting and I commend them for their effort but it doesn't represent a complete understanding of the Skype protocol to the extent that an Asterisk server could speak the Skype protocol. They say that much of the Skype protocol is encrypted and needs to be inferred to this point from the types and locations of messages that are being sent. MARK. Paul Hewlett wrote: On Monday 19 December 2005 14:23, Francesco Peeters (Asterisk) wrote: On Mon, December 19, 2005 11:33, Evert Meulie said: Hi all! I am aware of products like http://www.rsdevs.com/psgw_sip.shtml which act as a gateway, but what I'd really like is a for example an Asterisk module that can route calls to Skype, perhaps the same principle as IAX2? I'm assuming more people are interested in this, but... does it exist already? There is no such thing yes, and as Skype is closed source, it'll have to wait until someone reverse-engineers it... (Sniffing the protocol will be hard, as it is - supposedly - encrypted) 2 guys (Schulzrinne and Baset] of Columbia University have done it. See www1.cs.columbia.edu/~library/TR-repository/reports/reports-2004/cucs-039-04.pdf ___ --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] VoIPJet Support Contact
I'm all for criticism where it's due but I'm sure for all the bashing of Voipjet going on in this thread I'm sure there are just as many non-users who are generally happy with the service they provide and the price at which they provide it. I for one am also a customer of Verizon, a fact I'd rather not advertise in case anyone might get the false impression I am happy with the service they provide and the price at which they provide it. I don't think any of the VoIP wholesalers I deal with provide stellar customer service. Contrary to the bigger telco's, when you do finally get their attention they do their best to resolve your problem. Those that just really don't get it (remember LiveVoIP?) don't last. Otherwise, I think many of them are people like many of us who are trying to find a place in a difficult market. If you want wholesale termination/origination with an SLA attached then you're going to have to pay for it. MARK. Chris Mason (Lists) wrote: NON-DISCLOSURE: ALL CUSTOMERS USING VOIPJET'S SERVICE ARE SPECIFICALLY PROHIBITED FROM DISCLOSING TO OTHERS THAT THEY USE VOIPJET'S SERVICE, I use Voipjet, I have used Voipjet... Did I mention I use Voipjet? I'd like to teach the world to sing (about using Voipjet)... So sue me Voipjet, or better still, refund the outstanding balance so I can use it with a service that doesn't make people agree to stupid unenforcable rules. Another LiveVoip in the making. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detect alternate line in Broadvoice inbound context
Just as a followup, you need to be a bit careful and test this out if
you make any changes to your Broadvoice account. I added a second
virtual number and they switched the number that was previously
specifying a distinctive ring of Bellcore-dr3 to Bellcore-dr4. The last
number added now specifies Bellcore-dr3. Sending the number would be so
much more reliable...
MARK.
Mark Hulber wrote:
Ok, your solution does work but in looking at my console output I saw
that SIPGetHeader was deprecated for the new dialplan function
SIP_HEADER. Below is my modification. You don't need a priority+101.
exten = 212999,1,Set(Var_Alert=${SIP_HEADER(Alert-Info)})
exten = 212999,n,GotoIf($[${Var_Alert} =
http://127.0.0.1/Bellcore-dr3]?x-916999,1:x-212999,1)
In this case, the 212 number is the primary number.
Thanks,
MARK.
Samy Antoun wrote:
Mark,
1. Make sure that SIPGetHeader application is registered
CLI show application SIPGetHeader
if it is registered you'll get
-= Info about application 'SIPGetHeader' =-
[Synopsis]
Get a SIP header from an incoming call
[Description]
SIPGetHeader(var=headername):
Sets a channel variable to the content of a SIP header
Skips to priority+101 if header does not exist
Otherwise returns 0
If not,
Your application(s) is (are) not registered
If the application is not registered, I can't recommend anything for
you, I had an Asterisk system with ver 1.0 (no SIPGetHeader) and I
tried to patch it with any of the following with no luck:
http://bugs.digium.com/bug_view_page.php?bug_id=0002838
http://bugs.digium.com/view.php?id=2924
If you have it registered, here is a sample of my setup:
[bvdr]
exten = s,1,Answer exten = s,2,DigitTimeout(5) exten =
s,3,ResponseTimeout(10) exten =
s,4,SIPGetHeader(Var_Alert=Alert-Info) exten =
s,5,GotoIf($[${Var_Alert} =
http://127.0.0.1/Bellcore-dr3]?ext-local,320,1) exten =
s,6,Goto(ext-local,200,1)
This setup for ONE Distinctive Ring only (Bellcore-dr3), if you have
more than one, you can use sip debug to retrieve the header
information
The BEST reference for this subject is:
http://voxilla.com/PNphpBB2-viewtopic-t-3935-highlight-dring1.html
Hope this helps
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[Asterisk-Users] Detect alternate line in Broadvoice inbound context
I have a single Broadvoice account with more than one number. I am trying to distinguish between the numbers on an inbound call. I have already tried using different incoming extensions that match each number but it always defaults to the primary. Someone earlier mentioned SIPGetHeader as a possible solution but I'm not sure how that would work. The only field that might possibly contain the distinctive number that I can tell is list_route but I have never used SIPGetHeader and don't know if it even makes sense in this case. Does anyone have a solution for this? I also tried DNID with no luck. MARK. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detect alternate line in Broadvoice inbound context
Ok, your solution does work but in looking at my console output I saw that
SIPGetHeader was deprecated for the new dialplan function SIP_HEADER. Below is
my modification. You don't need a priority+101.
exten = 212999,1,Set(Var_Alert=${SIP_HEADER(Alert-Info)})
exten = 212999,n,GotoIf($[${Var_Alert} =
http://127.0.0.1/Bellcore-dr3]?x-916999,1:x-212999,1)
In this case, the 212 number is the primary number.
Thanks,
MARK.
Samy Antoun wrote:
Mark,
1. Make sure that SIPGetHeader application is registered
CLI show application SIPGetHeader
if it is registered you'll get
-= Info about application 'SIPGetHeader' =-
[Synopsis]
Get a SIP header from an incoming call
[Description]
SIPGetHeader(var=headername):
Sets a channel variable to the content of a SIP header
Skips to priority+101 if header does not exist
Otherwise returns 0
If not,
Your application(s) is (are) not registered
If the application is not registered, I can't recommend anything for
you, I had an Asterisk system with ver 1.0 (no SIPGetHeader) and I
tried to patch it with any of the following with no luck:
http://bugs.digium.com/bug_view_page.php?bug_id=0002838
http://bugs.digium.com/view.php?id=2924
If you have it registered, here is a sample of my setup:
[bvdr]
exten = s,1,Answer
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(10)
exten = s,4,SIPGetHeader(Var_Alert=Alert-Info)
exten = s,5,GotoIf($[${Var_Alert} =
http://127.0.0.1/Bellcore-dr3]?ext-local,320,1)
exten = s,6,Goto(ext-local,200,1)
This setup for ONE Distinctive Ring only (Bellcore-dr3), if you have
more than one, you can use sip debug to retrieve the header
information
The BEST reference for this subject is:
http://voxilla.com/PNphpBB2-viewtopic-t-3935-highlight-dring1.html
Hope this helps
__
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Re: [Asterisk-Users] Disa dialplan
You can pass a context to DISA for dialing outbound. Do you have a
dialplan that works like this for non-DISA calls? You can use the same
one. Otherwise, I do this with nested dialplans by putting the most
specific and longest rules first. By nesting, you only enter an
included context if none of the extensions are satisfied on the current
context. You'll have to ignore all the macros and just assume for the
most part that you would Dial there.
exten = ,n,DISA(no-password|dialout)
[dialout]
include = dialout2
include = check-voicemail
include = getparked
; Emergency
exten = 911,1,Goto(nineoneone,s,1)
; City Services
exten = 311,1,Set(CALLEDNUMBER=${EXTEN})
exten = 311,n,Macro(dialHC,Zap/3/${EXTEN},30)
; Information
exten = _1NXX5551212,1,Set(CALLEDNUMBER=${EXTEN})
exten = _1NXX5551212,n,Macro(dialHC,Zap/3/${EXTEN},30)
exten = _NXX5551212,1,Set(CALLEDNUMBER=${EXTEN})
exten = _NXX5551212,n,Macro(dialHC,Zap/3/1${EXTEN},30)
exten = _5551212,1,Set(CALLEDNUMBER=${EXTEN})
exten = _5551212,n,Macro(dialHC,Zap/3/${EXTEN},30)
exten = 411,1,Set(CALLEDNUMBER=${EXTEN})
exten = 411,n,Macro(dialHC,Zap/3/${EXTEN},30)
exten = 611,1,Set(CALLEDNUMBER=${EXTEN})
exten = 611,n,Macro(dialHC,Zap/3/${EXTEN},30)
; Timeout
exten = t,1,Playback(vm-goodbye)
exten = t,n,Macro(rhangup)
; Invalid Entry
exten = i,1,Macro(badentry,${INVALID_EXTEN})
exten = i,n,Macro(rhangup)
; Hangup
exten = h,1,Macro(rhangup)
[dialout2]
include = dialout3
include = onex
; Local
exten = _1212NXX,1,Macro(sddial-defid,${EXTEN})
exten = _212NXX,1,Macro(sddial-defid,1${EXTEN})
exten = _1646NXX,1,Macro(sddial-defid,${EXTEN})
exten = _646NXX,1,Macro(sddial-defid,1${EXTEN})
exten = _1917NXX,1,Macro(sddial-defid,${EXTEN})
exten = _917NXX,1,Macro(sddial-defid,1${EXTEN})
; Toll-Free
exten = _1800NXX,1,Macro(sddial-tf-defid,${EXTEN})
exten = _1866NXX,1,Macro(sddial-tf-defid,${EXTEN})
exten = _1877NXX,1,Macro(sddial-tf-defid,${EXTEN})
exten = _1880NXX,1,Macro(sddial-tf-defid,${EXTEN})
exten = _1881NXX,1,Macro(sddial-tf-defid,${EXTEN})
exten = _1882NXX,1,Macro(sddial-tf-defid,${EXTEN})
exten = _1888NXX,1,Macro(sddial-tf-defid,${EXTEN})
exten = _800NXX,1,Macro(sddial-tf-defid,1${EXTEN})
exten = _866NXX,1,Macro(sddial-tf-defid,1${EXTEN})
exten = _877NXX,1,Macro(sddial-tf-defid,1${EXTEN})
exten = _880NXX,1,Macro(sddial-tf-defid,1${EXTEN})
exten = _881NXX,1,Macro(sddial-tf-defid,1${EXTEN})
exten = _882NXX,1,Macro(sddial-tf-defid,1${EXTEN})
exten = _888NXX,1,Macro(sddial-tf-defid,1${EXTEN})
; Toronto
exten = _1289NXX,1,Macro(set647cid)
exten = _1289NXX,n,Macro(lddial-availCDN,${EXTEN:1})
exten = _289NXX,1,Macro(set647cid)
exten = _289NXX,n,Macro(lddial-availCDN,${EXTEN})
exten = _416NXX,1,Macro(set647cid)
exten = _416NXX,n,Macro(lddial-availCDN,${EXTEN})
exten = _1416NXX,1,Macro(set647cid)
exten = _1416NXX,n,Macro(lddial-availCDN,${EXTEN:1})
exten = _1647NXX,1,Macro(set647cid)
exten = _1647NXX,n,Macro(lddial-availCDN,${EXTEN:1})
exten = _647NXX,1,Macro(set647cid)
exten = _647NXX,n,Macro(lddial-availCDN,${EXTEN})
exten = _905NXX,1,Macro(set647cid)
exten = _905NXX,n,Macro(lddial-availCDN,${EXTEN})
exten = _1905NXX,1,Macro(set647cid)
exten = _1905NXX,n,Macro(lddial-availCDN,${EXTEN:1})
; California
exten = _415NXX,1,Macro(set916cid)
exten = _415NXX,n,Macro(lddial-avail,1${EXTEN})
exten = _1415NXX,1,Macro(set916cid)
exten = _1415NXX,n,Macro(lddial-avail,${EXTEN})
exten = _707NXX,1,Macro(set916cid)
exten = _707NXX,n,Macro(lddial-avail,1${EXTEN})
exten = _1707NXX,1,Macro(set916cid)
exten = _1707NXX,n,Macro(lddial-avail,${EXTEN})
exten = _1831NXX,1,Macro(set916cid)
exten = _1831NXX,n,Macro(lddial-avail,${EXTEN})
exten = _831NXX,1,Macro(set916cid)
exten = _831NXX,n,Macro(lddial-avail,1${EXTEN})
exten = _916NXX,1,Macro(set916cid)
exten = _916NXX,n,Macro(lddial-avail,1${EXTEN})
exten = _1916NXX,1,Macro(set916cid)
exten = _1916NXX,n,Macro(lddial-avail,${EXTEN})
; WestVirginia
exten = _304NXX,1,Macro(set304cid)
exten = _304NXX,n,Macro(lddial-avail,1${EXTEN})
exten = _1304NXX,1,Macro(set304cid)
exten = _1304NXX,n,Macro(lddial-avail,${EXTEN})
[dialout3]
#include ldrates.conf
include = onex
; Assume 212
exten = _NXX,1,Macro(sddial-defid,*821212${EXTEN})
**ldrates.conf**
[onex]
include = twox
[twox]
include = threex
[threex]
include = fourx
[fourx]
include = fivex
[fivex]
include = anyx
[anyx]
exten = _011.,1,Macro(setdefcallid)
exten = _011.,n,Macro(lddial-avail,${EXTEN})
exten = _NXXNXX,1,Macro(setdefcallid)
exten = _NXXNXX,n,Macro(lddial-avail,1${EXTEN})
exten =
Re: [Asterisk-Users] Moments of silence - take2
I'm not sure if a failed qualify will affect your active call but you might want to try to use the qualifysmoothing variable in iax.conf. This won't disqualify a peer for a single bad sample. ;qualify=yes; Make sure this peer is alive ;qualifysmoothing = yes ; use an average of the last two PONG ; results to reduce falsly detected LAGGED hosts ; Default: Off ;qualifyfreqok = 6 ; how frequently to ping the peer when ; everything seems to be ok, in milliseconds ;qualifyfreqnotok = 1 ; how frequently to ping the peer when it's ; either LAGGED or UNAVAILABLE, in milliseconds Adam Moffett wrote: I'm sorry, that previous message might have made more sense if it had all the information that I had intended to send. We are having moments of silence in the middle of phone calls. Generally it's not more than a few seconds, but it's still a nuisance. Our IAX providers (we have 2) become unreachable for periods of 5-15 seconds roughly 3 times an hour. It happens to both providers, but not at the same time. Below you'll find a log excerpt (cat messages | grep teliax) with regards to that. I was wondering if the two issues are related. Either way, does any one have any experience with regards to silence in the middle of phone calls? What possible causes should I be looking at. Nov 2 17:44:36 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now UNREACHABLE! Time: 68 Nov 2 17:44:46 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now REACHABLE! Time: 69 Nov 2 18:42:56 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now UNREACHABLE! Time: 68 Nov 2 18:43:06 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now REACHABLE! Time: 67 Nov 2 20:03:17 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now UNREACHABLE! Time: 68 Nov 2 20:03:27 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now REACHABLE! Time: 67 Nov 2 23:26:46 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now TOO LAGGED (2076 ms)! Nov 2 23:26:56 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now REACHABLE! Time: 68 Nov 2 23:44:01 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now UNREACHABLE! Time: 68 Nov 2 23:44:11 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now REACHABLE! Time: 68 Nov 2 23:58:16 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now UNREACHABLE! Time: 70 Nov 2 23:58:26 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now REACHABLE! Time: 73 Nov 3 02:58:48 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now UNREACHABLE! Time: 67 Nov 3 02:58:58 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now REACHABLE! Time: 69 Nov 3 06:24:19 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now UNREACHABLE! Time: 69 Nov 3 06:24:29 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now REACHABLE! Time: 68 Nov 3 06:29:31 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now TOO LAGGED (2086 ms)! Nov 3 06:29:42 NOTICE[16535] chan_iax2.c: Peer 'teliax' is now REACHABLE! Time: 69 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.2-beta2 odd CLI output
I think for SIP the control channel can still go through the proxy while the data is bridged natively allowing you to still account for the call. I'm not sure of the details on how Asterisk does it. MARK. David Bandel wrote: On 11/2/05, Mark Hulber [EMAIL PROTECTED] wrote: I think this means that it attempted to create a native bridge, which is that it was trying to have the call go directly between the two endpoints instead of going through the asterisk server but that process failed. So in that case, Asterisk continued to proxy the call data. If that's the case, a better output might have been, ... was unsuccessful, server will continue to bridge call, or something along those lines. MARK. Thanx, Mark. Makes sense since I deliberately put: canreinvite=no in the configuration of both SIP phones. Tough to account for calls that are not proxied. Ciao, David A. Bandel -- Focus on the dream, not the competition. - Nemesis Air Racing Team motto ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: musiconhold errors in 1.2.0-beta1
This is an old issue on which you can seach and find info. Some info indicates that you need a timing source such as a zaptel card or ztdummy. Other suggests that if you are using native music on hold and mpg123 is in the path you might run into this error. MARK. Daniel Corbe wrote: Has anyone run into this problem yet? -Daniel On 9/9/05, Daniel Corbe [EMAIL PROTECTED] wrote: I'm getting a FLOOD of these types of messages on my MAC OS X box: Sep 9 14:46:31 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! Sep 9 14:46:37 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! Sep 9 14:46:37 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! Sep 9 14:46:43 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! -Daniel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to call each other for dynamic ip hosts
As I understand it, you can initiate the call by having one of the dynamic endpoints call the other through the fixed ip host and then the fixed host can allow the two endpoints to create a native bridge. Otherwise, I think you'll have to somehow cache the registration at the dynamics hosts and try to make a direct call using that information. I don't really know if it's easy or hard to accomplish this. MARK. fun wrote: Hi, I have one host with fixed ip and two hosts with dynamic ip. These dynamic hosts should connect to the fixed ip host to register, so fixed and dynamic host can call each other without problem. My question is, how to let dynamic hosts can also call each other? (use iax2) Thanks for the help! Dominic ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Basic question...
It probably makes no difference to your problem but it's canreinvite not canreinvete. You'll want to include dialout extensions in [siptest]. For instance, maybe include your default context. MARK. Wagner Nunes wrote: Hi all!!! I have an asterisk compiled and started in one computer here at home, so I create 2 sip useres that request autentication to the asterisk using X-Lite.. The useers are log in all right, but when i try to have a call between they, it not work... I set the context as siptest, so what do i need to set in this context do make it work??? the sip.conf is down here... tkx all!!! [general] context=default svrlookup=yes [135140] type= friend secret=teste001 qualify=yes nat=no host=dynamic canreinvete=no context=siptest [135141] type=friend secret=teste qualify=yes nat=no host=dynamic canreinvete=no context=siptest Yahoo! Acesso Grátis: Internet rápida e grátis. Instale o discador agora! http://us.rd.yahoo.com/mail/br/tagline/discador/*http://br.acesso.yahoo.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap Polarity Reversal
Previously I would get two events on my Zap channel which indicated ringing and answered. Now I am getting polarity reversal events: Nov 2 07:01:25 NOTICE[18246]: chan_zap.c:6014 ss_thread: Got event 17 (Polarity Reversal)... Nov 2 07:01:28 NOTICE[18246]: chan_zap.c:6014 ss_thread: Got event 17 (Polarity Reversal)... I am using CVS Head from: Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-11-02 05:13:32 UTC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap Polarity Reversal
I am in the US, NYC using a TDM400 card. I never have never seen this issue until now. I see some code has been changed in this area recently. MARK. Rich Adamson wrote: Previously I would get two events on my Zap channel which indicated ringing and answered. Now I am getting polarity reversal events: Nov 2 07:01:25 NOTICE[18246]: chan_zap.c:6014 ss_thread: Got event 17 (Polarity Reversal)... Nov 2 07:01:28 NOTICE[18246]: chan_zap.c:6014 ss_thread: Got event 17 (Polarity Reversal)... I am using CVS Head from: Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-11-02 05:13:32 UTC ___ Sounds like a new feature. Have you tried reversing the polarity or putting a butt set that indicates pol on the line? I don't know that it makes a difference but you might as well correct it if it is reversed. That actually sounds more like whatever telco he's connecting to is providing answer supervision in the form of polarity reversal. Without knowing more about which country / telco, there is no way to tell. Note the polarity reversal is happening _after_ asterisk gets a call, therefore not likely to have anything to do with reversed tip/ring. That same message does not occur in the US with the analog TDM card, so not sure what the OP has or is connected to. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Warning -- chan_iax2.c: ast_sched_runq tasks
I've started to see this lately on outgoing IAX calls using CVS Head: -- IAX2/voipjet-out-5 is making progress passing it to SIP/64.26.157.252-094e4058 -- IAX2/voipjet-out-6 is making progress passing it to SIP/64.26.157.252-094e4058 Nov 2 16:56:15 WARNING[22566]: chan_iax2.c:7948 network_thread: chan_iax2: ast_sched_runq ran 2502 scheduled tasks all at once Nov 2 16:56:15 WARNING[22566]: chan_iax2.c:7948 network_thread: chan_iax2: ast_sched_runq ran 3131 scheduled tasks all at once Nov 2 16:56:15 WARNING[22566]: chan_iax2.c:7948 network_thread: chan_iax2: ast_sched_runq ran 3074 scheduled tasks all at once -- IAX2/voipjet-out-6 answered SIP/64.26.157.252-094e4058 Any interpretation? MARK. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Warning -- chan_iax2.c: ast_sched_runq tasks
Not busy at all. Like these were the only calls in progress at the time. MARK. Rich Adamson wrote: I've started to see this lately on outgoing IAX calls using CVS Head: -- IAX2/voipjet-out-5 is making progress passing it to SIP/64.26.157.252-094e4058 -- IAX2/voipjet-out-6 is making progress passing it to SIP/64.26.157.252-094e4058 Nov 2 16:56:15 WARNING[22566]: chan_iax2.c:7948 network_thread: chan_iax2: ast_sched_runq ran 2502 scheduled tasks all at once Nov 2 16:56:15 WARNING[22566]: chan_iax2.c:7948 network_thread: chan_iax2: ast_sched_runq ran 3131 scheduled tasks all at once Nov 2 16:56:15 WARNING[22566]: chan_iax2.c:7948 network_thread: chan_iax2: ast_sched_runq ran 3074 scheduled tasks all at once -- IAX2/voipjet-out-6 answered SIP/64.26.157.252-094e4058 Any interpretation? The above would suggest a _very_ busy box. Any idea how many calls were in progress during this? Some of the folks on the -dev list might be interested in seeing the above numbers, so I'll cross-post my response over there as well. Rich ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Warning -- chan_iax2.c: ast_sched_runq tasks
I can get it to repeat. It happens in a condition that has had no problem normally in the past. I have an incoming call, SIP or ZAP it doesn't really matter and then I'm dialing out on IAX to try and connect the call to one or more numbers on an IAX channel. For each IAX channel I create I seem to be getting at least one of these messages: -- SIP/sipura2_1-085a is ringing -- SIP/sipura2_2-0f28 is ringing -- Call accepted by 64.34.45.100 (format ulaw) -- Format for call is ulaw -- IAX2/voipjet-out-5 is making progress passing it to SIP/147.135.20.128-08d36c60 Nov 2 18:01:06 WARNING[2781]: chan_iax2.c:7948 network_thread: chan_iax2: ast_sched_runq ran 3271 scheduled tasks all at once -- Hungup 'Zap/3-1' In this second case above I was only dialing out to a single number. In the previous case below I was dialing out to two numbers. Here's what I am running. I only started seeing it recently: Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-11-02 22:22:26 UTC MARK. Rich Adamson wrote: I've started to see this lately on outgoing IAX calls using CVS Head: -- IAX2/voipjet-out-5 is making progress passing it to SIP/64.26.157.252-094e4058 -- IAX2/voipjet-out-6 is making progress passing it to SIP/64.26.157.252-094e4058 Nov 2 16:56:15 WARNING[22566]: chan_iax2.c:7948 network_thread: chan_iax2: ast_sched_runq ran 2502 scheduled tasks all at once Nov 2 16:56:15 WARNING[22566]: chan_iax2.c:7948 network_thread: chan_iax2: ast_sched_runq ran 3131 scheduled tasks all at once Nov 2 16:56:15 WARNING[22566]: chan_iax2.c:7948 network_thread: chan_iax2: ast_sched_runq ran 3074 scheduled tasks all at once -- IAX2/voipjet-out-6 answered SIP/64.26.157.252-094e4058 Any interpretation? The above would suggest a _very_ busy box. Any idea how many calls were in progress during this? Some of the folks on the -dev list might be interested in seeing the above numbers, so I'll cross-post my response over there as well. Rich ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Warning -- chan_iax2.c: ast_sched_runq tasks
Ok. It seems my calls are completing ok. MARK. Rich Adamson wrote: Steve committed the source code changes to cvs-head to display these numbers, and I believe he was working on jitterbuffer or codec stuff. He has since posted an item that he wished he would not have done that as its generating lots of questions. His thought process was something along the lines that if too many tasks are in the secheduler, then servicing the code he was playing with might not occur in the time frames that one might expect. So he dumped those lines of code in there to find out. Since there is no other data, no one knows what those other scheduled tasks might be, so there's no way to answer the question of 'what's going on'. Just ignore the item for now. Rich From: Mark Hulber [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Warning -- chan_iax2.c: ast_sched_runq tasks Date: Wed, 02 Nov 2005 17:50:28 -0500 To: Rich Adamson [EMAIL PROTECTED] Not busy at all. Like these were the only calls in progress at the time. MARK. Rich Adamson wrote: I've started to see this lately on outgoing IAX calls using CVS Head: -- IAX2/voipjet-out-5 is making progress passing it to SIP/64.26.157.252-094e4058 -- IAX2/voipjet-out-6 is making progress passing it to SIP/64.26.157.252-094e4058 Nov 2 16:56:15 WARNING[22566]: chan_iax2.c:7948 network_thread: chan_iax2: ast_sched_runq ran 2502 scheduled tasks all at once Nov 2 16:56:15 WARNING[22566]: chan_iax2.c:7948 network_thread: chan_iax2: ast_sched_runq ran 3131 scheduled tasks all at once Nov 2 16:56:15 WARNING[22566]: chan_iax2.c:7948 network_thread: chan_iax2: ast_sched_runq ran 3074 scheduled tasks all at once -- IAX2/voipjet-out-6 answered SIP/64.26.157.252-094e4058 Any interpretation? The above would suggest a _very_ busy box. Any idea how many calls were in progress during this? Some of the folks on the -dev list might be interested in seeing the above numbers, so I'll cross-post my response over there as well. Rich ---End of Original Message- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.2-beta2 odd CLI output
I think this means that it attempted to create a native bridge, which is that it was trying to have the call go directly between the two endpoints instead of going through the asterisk server but that process failed. So in that case, Asterisk continued to proxy the call data. If that's the case, a better output might have been, ... was unsuccessful, server will continue to bridge call, or something along those lines. MARK. David Bandel wrote: Folks, Just cleared a call and had this on the CLI: -- Executing Macro(SIP/david-75e3, voicemail|sip/craigo) in new stack -- Executing Dial(SIP/david-75e3, sip/craigo|20|m) in new stack -- Called craigo -- Started music on hold, class 'default', on channel 'SIP/david-75e3' -- SIP/craigo-88db is ringing -- SIP/craigo-88db answered SIP/david-75e3 -- Stopped music on hold on SIP/david-75e3 -- Attempting native bridge of SIP/david-75e3 and SIP/craigo-88db -- Native bridge of SIP/david-75e3 and SIP/craigo-88db was unsuccessful == Refreshing DNS lookups. == Spawn extension (macro-voicemail, s, 1) exited non-zero on 'SIP/david-75e3' in macro 'voicemail' == Spawn extension (longdistance, 1001, 1) exited non-zero on 'SIP/david-75e3' Odd thing is this call was successful. We talked for about 10 minutes. So why would the SIP-SIP native bridge say UNsuccessful? David A. Bandel -- Focus on the dream, not the competition. - Nemesis Air Racing Team motto ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID to SIPURA-1000, 2000, 3000 Handset Prlblem in only showing the destination callerId
I don't use quotes on either if that makes a difference. When are you
setting it? Maybe you are losing the incoming number by the time you
set it the number has been changed to the extension. I use Sipuras and
have no problem with this. Here's an example of a macro I use when
forwarding an incoming call. I call this macro prior to dialing to
forward. The second part of this lets the receiver know that there
wasn't any callerid on the incoming call but that Asterisk is forwarding
the call to them.
[macro-fwdcallid]
exten = s,1,GotoIf(${CALLERIDNUM}?known:set)
exten = s,n(known),Set(CALLERID(number)=${CALLERIDNUM})
exten = s,n,Set(CALLERID(name)=${CALLERIDNAME})
exten = s,s+2(set),Set(CALLERID(number)=212000|a)
exten = s,n,Set(CALLERID(name)=Anonymous Forward|a)
MARK.
Ben Higley wrote:
Is there a resolution to this problem. It was posted a few weeks back.
But just chiming in again to see if someone has had any luck:
Problem:
Incoming call to a Sipura 2000, 1000, 3000 ATA.
I use the SetCallerID(name)=blah blah blah
SetCAllerId(number)=1234567890
However, On the handset, in this case, it's extension 2000. I see on the
handset display
Blah Blah Blah
2000
I have tried in the dial command to use the o - as shown in the show
application dial - to use old style, but this has not solved the issue.
This makes the end user not able to go through the caller id missed calls
to dial that person back...
Thanks...
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Re: [Asterisk-Users] Timestamps in Console?
Yes, or this for example:
[macro-rhangup]
exten = s,1,NoOp(DIALSTATUS=${DIALSTATUS})
exten = s,n,NoOp(TIME=${DATETIME})
exten = s,n,Hangup
I also output the date and time prior to dialing out.
MARK.
Sherwood McGowan wrote:
You could always just add some
exten = NUM,PRIO,VERBOSE(LEVEL|${DATETIME} -- THIS EVENT HAPPENED)
type commands in your dialplan to force output of the date time, and
you can even reduce the amount of verbosity to the CLI by using it
liberally to signify events, so you don't have to watch EVERYTHING.
Sherwod
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of
[EMAIL PROTECTED]
*Sent:* Monday, October 31, 2005 9:31 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [Asterisk-Users] Timestamps in Console?
Hello!
Lately, I've been keeping a close eye on an Asterisk box by
staying logged into the console for long periods of time.
However, it can be very difficult to know how long a telephone
call lasts when this is all you see:
-- Executing Dial(SIP/SIP105-8e34, Zap/g2/Number|60|t) in
new stack
-- Called g2/Number
-- Zap/5-1 answered SIP/SIP105-8e34
-- Hungup 'Zap/5-1'
Did that telephone call last only a few seconds because there was
a problem, or a few minutes because there wasn't? It's impossible
to tell.
Is there a way to add timestamps to each line in the console so
you know exactly how long a call took? Or is there another way of
telling directly within the console?
Thank you very much!
Tim Massey
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Re: [Asterisk-Users] no sip peers after restarting asterisk?
I noted this on Friday. I don't think I had problems using the devices but sip show peers took some time to show the registrations. MARK. Kevin P. Fleming wrote: Rich Adamson wrote: Once the phones register again, they can be called, but not until then. Not sure what's going on yet... anyone seeing the same? Mark just committed some changes in this area... it's possible those changes unintentionally broke writing of SIP registry information into the astdb. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Top and asterisk performance
It might be helpful to show what's using the CPU (the rest of Top). MARK. Julian Lyndon-Smith wrote: We had to move from a old * server to a new one in a hurry (hardware failure). The old server was a dual pentium 700 with 512MB ram running fedora core 2, the new one is a single 3GHz Pentium with 1gb ram. The same number of people are connected to the new server as the old, the same number of inbound calls to the isdn30 etc (on average 20 calls active at any time (SIP and ZAP)). Basically, just a server swapout. I must be reading top wrong, because the old server had a idle of approx 30%, whereas the new server is top - 13:35:21 up 12 days, 23:57, 1 user, load average: 7.11, 7.20, 7.21 Tasks: 98 total, 9 running, 89 sleeping, 0 stopped, 0 zombie Cpu(s): 99.0% us, 1.0% sy, 0.0% ni, 0.0% id, 0.0% wa, 0.0% hi, 0.0% si Mem: 1034640k total, 144792k used, 889848k free,21952k buffers Swap: 2031608k total,0k used, 2031608k free,61248k cached Notice the 99.0% us. This fluctuates between 80 and 99%. The other difference is that the new server is on cvs-head as of today - I did say that it was an emergency :) whereas the old server was cvs-head from june sometime. Is it just me, or is there a problem ? Julian. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Host Unspecified
In recent CVS Head build when I run: sip show peers my dynamic peers show: Name/username HostDyn Nat ACL Port Status sipura2_2/sipura2_2(Unspecified)D N 0UNKNOWN sipura2_1/sipura2_1(Unspecified)D N 0UNKNOWN sipura1_2/sipura1_2(Unspecified)D 0UNKNOWN sipura1_1/sipura1_1(Unspecified)D 0UNKNOWN So I can no longer see which IP they are at and I can't see the qualify times. If this the new designed behavior? Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-10-28 19:30:31 UTC MARK. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where MeetMe application
/usr/src/asterisk/apps/app_meetme.so /usr/lib/asterisk/modules/app_meetme.so Fabio Montemaggiore wrote: I haven't app_meetme.so file... Where I can search? ___ Yahoo! Messenger: chiamate gratuite in tutto il mondo http://it.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extended SIP registration failures
I had some complaints today that one of my incoming SIP numbers was failing for several hours. I looked at my console and didn't see anything unusual but SIP show registry confirmed that my registrations were in a failed state. I did a SIP reload and saw this in the output: Sep 22 18:54:48 NOTICE[9997]: sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule entry 312896! Sep 22 18:54:48 NOTICE[9997]: sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule entry 313687! After the reload the registrations were successful. Without knowing much about it, apparently they were scheduled to retry but that failed for some reason. Any ideas? MARK. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
