RE: [Asterisk-Users] Dutch SIP or IAX numbers
Hi, > -Original Message- > How knows where I can get a Dutchphone number for asterisk? > > Pilmo is not delivering one for home use. I think you are physically outside the netherlands, right ? Would you care for an 087 number ? Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 - Phone TIme
Paul Hales wrote: It now works - but only in the latest (1.5+) firmware releases. Where are the 1.5 releases? I see only 1.4.1 on all the Polycom sites. Regards, Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with ser to share the load
Hello Deepak, 1. don't post multiple times. it's annoying. enough said. 2. run asterisk in verbose mode (start it with asterisk -vgc), place a call from a SIP endpoint behind SER to the asterisk server, and see what happens in the asterisk CLI. 3. if you don't see anything there, get ngrep and place a call from the SIP endpoint while running "ngrep SIP" and post the output. 4. are asterisk and SER on the same machine? 5. if all else fails put autocreatepeer=yes in your sip.conf - this has bad security consequences, but it is useful for debugging. -yair On 12/2/04, Deepak Dhiman <[EMAIL PROTECTED]> wrote: > > > Hi friends ! > > Can anybody help me to configure asterisk with ser so that I can share > the load of the asterisk with ser server. I have tried it but my > asterisk is not showing registrations of the user agent, as given in the > asterisk wiki/asterisk+at+large. I don't know what is the problem, but > can assure abt the ser that is is running well and also forwarding > packets to asterisk server but * is not getting these packets. Can > anybody tell me that what`s wrong with my Asterisk server? Do I need to > change /add something in sip.conf? Please help me . > > Regards, > > Deepak Dhiman > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with ser to share the load
Hi friends ! Can anybody help me to configure asterisk with ser so that I can share the load of the asterisk with ser server. I have tried it but my asterisk is not showing registrations of the user agent, as given in the asterisk wiki/asterisk+at+large. I don't know what is the problem, but can assure abt the ser that is is running well and also forwarding packets to asterisk server but * is not getting these packets. Can anybody tell me that what`s wrong with my Asterisk server? Do I need to change /add something in sip.conf? Please help me . Regards, Deepak Dhiman ___ 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] Problem with X101P
Scott Stingel wrote: >Some questions: > >What country are you in? > >Is there anything else connected to the line from the PSTN? It sounds >like you have a marginal condition, such as insufficient loop current >perhaps. > >Do have any features, such as call waiting, on the line? > >Do you know how far you are from the central office? > >Do you have another line you can switch to and try the same card? > >Does the Red alarm occur at the moment the call is disconnected, or >afterward? Sorry for late reply. Answers: --I am in Bangladesh. --No there is nothing else connected with my PSTN line. But, in future that line would be connected with the Fax simultaneously. -- No call waiting feature on the line. But in zapata configuration this feature is true. -- Sorry I didn't get that question. --Yeah I have other two lines and I have checked with those lines with the same card. Same thing happens for those. -- The Red Alarm occurs when I am connected and having conversation with other party. Thank you ___ 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: RE: [Asterisk-Users] Problems with TDM400P card
Thanks for the info. What hard drives are you using ide or serial ata. Does it make a difference. Thanks Geoffrey Sachs - Original Message - From: "Anton Krall" <[EMAIL PROTECTED]> To: "'Kim Culhan'" <[EMAIL PROTECTED]>; "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Saturday, April 30, 2005 7:44 AM Subject: RE: RE: [Asterisk-Users] Problems with TDM400P card Hows does this look? Opened pseudo zap interface, measuring accuracy... 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8194 sample intervals 99.975586% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8193 sample intervals 99.987793% --- Results after 13 passes --- Best: 100.00 -- Worst: 99.975586 -- Average: 99.987793 Good enough and what do I need to check in order to make 100%? What does the test actually measure? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Kim Culhan |Sent: Sábado, 30 de Abril de 2005 08:45 a.m. |To: asterisk-users@lists.digium.com |Subject: Re: RE: [Asterisk-Users] Problems with TDM400P card | |On Fri, April 29, 2005 4:58 pm, GEOFFREY SACHS said: |> I would also be interested in alternatives to the Tdm400p. I |have had |> endless problems with a tdm400p card not being able to get |the zttest |> numbers above |> 99.975 and as a result not being able eliminate an |intermitent but consistent echo. |> I have tried to date 4 different motherboard and hardware |combinations |> as well as different linux versions to no avial.I would |welcome some feedback on this. | |Since there appear to be several combinations of hardware and |operating system which don't work well, here is a combination |which appears to work fairly well: | |Intel 925XCV mb | |P-4 560 (3.6 gHz) | |wcfxs0: | |FreeBSD 5.4-STABLE | |zttest -v |Opened pseudo zap interface, measuring accuracy... | |8192 samples in 8192 sample intervals 100.00% |8192 samples in 8192 sample intervals 100.00% |8192 samples in 8192 sample intervals 100.00% |8192 samples in 8192 sample intervals 100.00% |8192 samples in 8192 sample intervals 100.00% |8192 samples in 8192 sample intervals 100.00% |8192 samples in 8192 sample intervals 100.00% |8192 samples in 8192 sample intervals 100.00% |8192 samples in 8192 sample intervals 100.00% |8192 samples in 8192 sample intervals 100.00% ^C |--- Results after 10 passes --- |Best: 100.00 -- Worst: 100.00 -- Average: 100.00 | |hope this helps | |-kim | |-- |[EMAIL PROTECTED] |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] Sipura SPA2000 dialplan vs Asterisk dialplan
I've got a Sipura SPA2000 ATA basically working (I can place calls between the extensions plugged into each of its ports) and part of that was setting up the dial plan on the SPA2000 to match the one in Asterisk. This seems like a pain to deal with long term and I don't know what exactly the dial plan built into the SPA2000 does for me over the one in Asterisk. Is there a way to disable the use of the SPA2000 dialplan so I don't have to keep it in synch? Or is there some reason why it would be a bad idea for me to do so? Thanks Steve ___ 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 Hears Ring During Attended Transfer?
Eric Wieling aka ManxPower wrote: Daryll Strauss wrote: I'm running Asterisk 1.0.7 and SPA-3000 Firmware 2.0.13g. A call comes in to my asterisk box via SIP (the Sipura isn't involved) and I answer it using an analog phone on the Sipura. I then decide to forward it to another phone. I flash the line, dial the new extension, and flash again. At this point I and the caller hear the destination phone ring. If I then hang up, the caller stops hearing the ring. Eventually the voicemail picks up and the caller hears the announcement and everything works as normal. It's just rather disconcerting for the caller to not hear anything. Is this an Asterisk bug, Sipura bug, or operator (me) error? Happens with Polycom phones as well. It may be related to this message when a transfer happens: Unable to handle indication 3 for 'SIP/0004f201e4b3-a-682d' ___ 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 Hears Ring During Attended Transfer?
Daryll Strauss wrote: I'm running Asterisk 1.0.7 and SPA-3000 Firmware 2.0.13g. A call comes in to my asterisk box via SIP (the Sipura isn't involved) and I answer it using an analog phone on the Sipura. I then decide to forward it to another phone. I flash the line, dial the new extension, and flash again. At this point I and the caller hear the destination phone ring. If I then hang up, the caller stops hearing the ring. Eventually the voicemail picks up and the caller hears the announcement and everything works as normal. It's just rather disconcerting for the caller to not hear anything. Is this an Asterisk bug, Sipura bug, or operator (me) error? This is a known (by at least some people) problem. I don't know of a fix or a bug number. ___ 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] Caller Hears Ring During Attended Transfer?
I'm running Asterisk 1.0.7 and SPA-3000 Firmware 2.0.13g. A call comes in to my asterisk box via SIP (the Sipura isn't involved) and I answer it using an analog phone on the Sipura. I then decide to forward it to another phone. I flash the line, dial the new extension, and flash again. At this point I and the caller hear the destination phone ring. If I then hang up, the caller stops hearing the ring. Eventually the voicemail picks up and the caller hears the announcement and everything works as normal. It's just rather disconcerting for the caller to not hear anything. Is this an Asterisk bug, Sipura bug, or operator (me) error? - |Daryll ___ 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] which port is used when "asterisk -r "
Asterisk -r is used to reconnect to CLI of a running Asterisk system from the console, so there is no TCP port in use to do that. You sould SSH to your Asterisk server and "asterisk -r" to interact with the running Asterisk application. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk guy Sent: Sunday, May 01, 2005 7:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] which port is used when "asterisk -r " which TCP port is used when "asterisk -r " ? is there a command to connect to a remote machine ? ( asterisk -r remote-machine-ip ?) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] which port is used when "asterisk -r "
which TCP port is used when "asterisk -r " ? is there a command to connect to a remote machine ? ( asterisk -r remote-machine-ip ?) ___ 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] how to disconnect a call manually
1 after giving command oh323 show channels, i want to disconnect a call, is there any command to disconnect a call? 2 how asterisk kill a hung/dead call ? for most commercial softswitch, there are a setting for maximum duration for a call. they will hang up it l if its duration reachs the limit. ___ 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] Problem with Sangoma/Adtran 600 installation
the same you want it to be linked to in zaptel.conf so if you config it as span1 in zaptel.conf it has to say 1 On 5/1/05, Chris Mason (Lists) <[EMAIL PROTECTED]> wrote: > x lk x > x x Interface Name-> w1g1x x > x x Operation Mode-> TDM_VOICE x x > x x TDM Voice Span-> 1 x x > x x Override Asterisk Echo Enable -> No > > What should the TDM Voice Span be? > > Chris Mason > www.anguillaguide.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 > -- Michal Bielicki http://www.aefirion.org/ http://www.asterisk.com.pl/ ___ 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] Set up SIP, now I'm getting a busy tone and weird (to me)messages on the console
--- Tim Connolly <[EMAIL PROTECTED]> wrote: > Is NAT=yes on, are you behind a firewall? Give us > some connectivity details. > Usually when you see maximum retries, its because > you have one-way > communications with the far end for some reason. Are > you setting "externip" > statically? To answer your questions, yes, I am behind a firewall. The asterisk server is the only device connected to a cheapo Netgear 4-port router/firewall. I'm not setting externip myself, so whatever the default is, it's getting used. I'm also NOT making outgoing calls, and there are no actual SIP devices attached ... I'm just trying to receive incoming calls forwarded from a different provider via SIP. Here is a complete sip.conf file ... do I need to provide anything else? sip.conf: ; ; SIP Configuration for Asterisk ; ; Syntax for specifying a SIP device in extensions.conf is ; SIP/devicename where devicename is defined in a section below. ; ; You may also use ; SIP/[EMAIL PROTECTED] to call any SIP user on the Internet ; (Don't forget to enable DNS SRV records if you want to use this) ; ; If you define a SIP proxy as a peer below, you may call ; SIP/proxyhostname/user or SIP/[EMAIL PROTECTED] ; where the proxyhostname is defined in a section below ; ; Useful CLI commands to check peers/users: ; sip show peers Show all SIP peers (including friends) ; sip show users Show all SIP users (including friends) ; sip show registry Show status of hosts we register with ; ; sip debug Show all SIP messages ; [general] ;context=default; Default context for incoming calls context=unwelcome-calls ; Default context for incoming calls ; After all, we don't want any random ; incoming calls to have access to outbound ; calling ;recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) ;realm=mydomain.tld ; Realm for digest authentication ; defaults to "asterisk" ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet ;pedantic=yes ; Enable slow, pedantic checking for Pingtel ; and multiline formatted headers for strict ; SIP compatibility (defaults to "no") ;tos=184; Set IP QoS to either a keyword or numeric val ;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY ;videosupport=yes ; Turn on support for SIP video ;disallow=all ; First disallow all codecs ;allow=ulaw ; Allow codecs in order of preference ;allow=ilbc ; Note: codec order is respected only in [general] ;musicclass=default ; Sets the default music on hold class for all SIP calls ; This may also be set for individual users/peers ;language=en; Default language setting for all users/peers ; This may also be set for individual users/peers ;relaxdtmf=yes ; Relax dtmf handling ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity ; when we're not on hold ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity ; when we're on hold (must be > rtptimeout) ;trustrpid = no ; If Remote-Party-ID should be trusted ;progressinband=no ; If we should generate in-band ringing always ;useragent=Asterisk PBX ; Allows you to change the user agent string ;nat=no ; NAT settings ; yes = Always ignore info and assume NAT ; no = Use NAT mode only according to RFC3581 ; never = Never attempt NAT mod
RE: [Asterisk-Users] Programing a call forward feature to cel phones
:) go ahead and send me a copy when you are done :) My gains are around 6 so you might be right... Maybe lowering them to 3 or so... |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Matt Riddell |Sent: Domingo, 01 de Mayo de 2005 08:13 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Programing a call forward |feature to cel phones | |What have you got for rxgain and txgain for the channels that |are going out over the PSTN? | |It sounds to me like you have echo above 0dB which would mean |that it would get louder with each iteration. | |This could be achieved by having bad echo as well as having |the gains set too high. | |Cool sound file! Mind if I use it in an industrial techno song? | |:) | |-- |Cheers, | |Matt Riddell |___ | |http://www.sineapps.com/news.php (Daily Asterisk News - html) |http://www.sineapps.com/rssfeed.php (Daily Asterisk News - |rss) ___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@Home bug
We installed AAH .05, tweaked it and learned more about dialplans, Queues (not included in that version of AMP), "upgrading" to CVS Head (needed atxfer/automon) and anything else we needed to scale AAH to our needs (75 agents 15k+ calls/day)than I believe we would have learned by simply trying to start with asterisk. AAH does a great job of showing people new to * what it can do once it's all put together. While I agree that AAH questions tend to be simple in nature and in some cases, as this one, not related to * mainstream installations, most of the questions deal with AMP, zaptel, libpri, spandsp, .conf files, etc. Please try not to immediately dismiss AAH questions to the forums simply because they are prefixed by "need help AAH". Now as for this question in particular, come on guys do a bit of leg work >From install_addon.sh (added in .9) echo " ---" echo "|Installing RS-232 console on COM1 |" echo " ---" echo "" echo "" echo "" echo "" if ! grep ttyS0 /etc/securetty >/dev/null 2>&1; then echo "s0:12345:respawn:/sbin/agetty -i -h -L 9600 ttyS0 vt100" >> /etc/inittab echo ttyS0 >> /etc/securetty fi Try downloading the tarball and looking at what it is that you are blindly installing on your system everytime you download the iso and burn it. /rant off Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Saturday, April 30, 2005 6:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] bug This is not that AAH mailing list, check out the fourms. On Sat, 30 Apr 2005, Manny A. Wise wrote: > After installation of [EMAIL PROTECTED] v1, I have an annoying message in > the screen, anyone know how to fix it > > > > INIT: Id "s0" respawning too fast: disable for 5 minutes > > > > Thanks > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] post-dial variable for whoever answered?
Tim Connolly wrote: "accounting-${TIMESTAMP}-${CALLERIDNUM}". I'd like to do a changemonitor() to make the filename "accounting-${TIMESTAMP}-${CALLERIDNUM}-" where is the extension number of the SIP client who actually answered. Keep in mind, its dialing multiple destinations, so I don't know what number picks up beforehand. From doc/README.variables: ${BRIDGEPEER}bridged peer This will contain the channel name of the channel that answered the call; it will not contain an 'extension number', as there is no such thing at point (since you could have dialed a peer that doesn't even _have_ an extension number in your dialplan). ___ 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] Set up SIP, now I'm getting a busy tone and weird (to me)messages on the console
Is NAT=yes on, are you behind a firewall? Give us some connectivity details. Usually when you see maximum retries, its because you have one-way communications with the far end for some reason. Are you setting "externip" statically? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of beonice Sent: Sunday, May 01, 2005 8:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Set up SIP,now I'm getting a busy tone and weird (to me)messages on the console Folks, I'm hoping someone has already run into this ... the only other complaint I've seen is here: http://lists.digium.com/pipermail/asterisk-bsd/2005-March/000640.html and that basically was a problem with the /etc/hosts ... my server is definitely described in my hosts file. I've been using asterisk with IAX and a voicepulse connect number. No problems at all receiving calls. Now, I've just purchased a DID in Canada from another provider, and their proxy only supports SIP. So, following the generic instructions I've found off the web, I set up my SIP.conf to point to voicepulse's server, and set up the other DID to point into this newly defined sip context, i.e., to uid:[EMAIL PROTECTED]/888 The problem? The remote DID, when called, simply gives me a busy signal. Also, on the asterisk console, I'm seeing these messages that don't tell me anything: May 1 18:37:09 WARNING[12065]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (Critical Request) May 1 18:37:23 NOTICE[12065]: chan_sip.c:4036 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again May 1 18:37:23 DEBUG[12065]: chan_sip.c:4150 transmit_register: Scheduled a registration timeout # 5 May 1 18:37:29 WARNING[12065]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 104 (Critical Request) May 1 18:37:43 NOTICE[12065]: chan_sip.c:4036 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again May 1 18:37:43 DEBUG[12065]: chan_sip.c:4150 transmit_register: Scheduled a registration timeout # 7 It looks like the remote DID is failing to register with the voicepulse server. Any hints on what could be the problem? If it helps, here is the relevant portion of my sip.conf file. [general] ;context=default; Default context for incoming calls context=unwelcome-calls ; Default context for incoming calls ; After all, we don't want any random ; incoming calls to have access to outbound ; calling - Maya Kurup, May 1, 2005 ;recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) ;realm=mydomain.tld ; Realm for digest authentication ; defaults to "asterisk" ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) ... register => uid:[EMAIL PROTECTED] ; We need to allow at least incoming calls to ; accept calls via libretel, etc. ; So, let's add a context for that: [888]; For incoming calls ONLY type=user ; This device takes incoming calls username=uid ; Username on device secret=secret ; Password for device host=srvr.voicepulse.com ; This host will not ; change frequently context=allowed_context ; Inbound calls from ; this host go ; to the normal context - and I have allowed_context described in my extensions.conf, it's the same one I'm using for regular IAX incoming calls, and works fine. The context for unwelcome-calls is as follows: [unwelcome-calls] ; ; Take unknown callers that may have found ; our system, and send them to a re-order tone. ; The string "_." matches any dialed sequence, so all ; calls will result in the Congestion tone application ; being called. They'll get bored and hang up eventually. ; exten => _.,1,Congestion --- Any help would be appreciated. Thanks, Maya __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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-
RE: [Asterisk-Users] Dutch SIP or IAX numbers
Thanks, But a small correction Dutchphone works only with one addpack modem and can only work with asterisk in combination with a X100P interface. It is the same solution as pilmo. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Sunday, May 01, 2005 2:01 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Dutch SIP or IAX numbers On 12:23, Sun 01 May 05, Asterisk wrote: > How knows where I can get a Dutchphone number for asterisk? > > Pilmo is not delivering one for home use. Three I use: http://www.speakup.nl (for now only busines accounts) http://www.talkin2ya.nl (both prepaid and postpaid) http://www.dutchphone.nl (only postpaid) -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence." ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Set up SIP, now I'm getting a busy tone and weird (to me) messages on the console
Folks, I'm hoping someone has already run into this ... the only other complaint I've seen is here: http://lists.digium.com/pipermail/asterisk-bsd/2005-March/000640.html and that basically was a problem with the /etc/hosts ... my server is definitely described in my hosts file. I've been using asterisk with IAX and a voicepulse connect number. No problems at all receiving calls. Now, I've just purchased a DID in Canada from another provider, and their proxy only supports SIP. So, following the generic instructions I've found off the web, I set up my SIP.conf to point to voicepulse's server, and set up the other DID to point into this newly defined sip context, i.e., to uid:[EMAIL PROTECTED]/888 The problem? The remote DID, when called, simply gives me a busy signal. Also, on the asterisk console, I'm seeing these messages that don't tell me anything: May 1 18:37:09 WARNING[12065]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (Critical Request) May 1 18:37:23 NOTICE[12065]: chan_sip.c:4036 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again May 1 18:37:23 DEBUG[12065]: chan_sip.c:4150 transmit_register: Scheduled a registration timeout # 5 May 1 18:37:29 WARNING[12065]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 104 (Critical Request) May 1 18:37:43 NOTICE[12065]: chan_sip.c:4036 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again May 1 18:37:43 DEBUG[12065]: chan_sip.c:4150 transmit_register: Scheduled a registration timeout # 7 It looks like the remote DID is failing to register with the voicepulse server. Any hints on what could be the problem? If it helps, here is the relevant portion of my sip.conf file. [general] ;context=default; Default context for incoming calls context=unwelcome-calls ; Default context for incoming calls ; After all, we don't want any random ; incoming calls to have access to outbound ; calling - Maya Kurup, May 1, 2005 ;recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) ;realm=mydomain.tld ; Realm for digest authentication ; defaults to "asterisk" ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) ... register => uid:[EMAIL PROTECTED] ; We need to allow at least incoming calls to ; accept calls via libretel, etc. ; So, let's add a context for that: [888]; For incoming calls ONLY type=user ; This device takes incoming calls username=uid ; Username on device secret=secret ; Password for device host=srvr.voicepulse.com ; This host will not ; change frequently context=allowed_context ; Inbound calls from ; this host go ; to the normal context - and I have allowed_context described in my extensions.conf, it's the same one I'm using for regular IAX incoming calls, and works fine. The context for unwelcome-calls is as follows: [unwelcome-calls] ; ; Take unknown callers that may have found ; our system, and send them to a re-order tone. ; The string "_." matches any dialed sequence, so all ; calls will result in the Congestion tone application ; being called. They'll get bored and hang up eventually. ; exten => _.,1,Congestion --- Any help would be appreciated. Thanks, Maya __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] post-dial variable for whoever answered?
I’m looking for suggestions on how to use changemonitor() to modify the filename monitor() is writing to, once an extension has picked up the call. For instance, the filename is currently “accounting-${TIMESTAMP}-${CALLERIDNUM}”. I’d like to do a changemonitor() to make the filename “accounting-${TIMESTAMP}-${CALLERIDNUM}-” where is the extension number of the SIP client who actually answered. Keep in mind, its dialing multiple destinations, so I don’t know what number picks up beforehand. Thanks Tim ___ 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] Programing a call forward feature to cel phones
What have you got for rxgain and txgain for the channels that are going out over the PSTN? It sounds to me like you have echo above 0dB which would mean that it would get louder with each iteration. This could be achieved by having bad echo as well as having the gains set too high. Cool sound file! Mind if I use it in an industrial techno song? :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] asterisk and USRobotics Courier V.Everything
hi for all, would like to know if asterisk supports the modem USRobotics Courier V.Everything. Looking for very I found one link that it says on a called module chan_modem_usr2976.so that the principle would function, but lowering the sources of asterisk I did not find this module, somebody can help me? thanks for all Guilherme ___ 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] Pre-Parse Extensions.conf?
Matthew Boehm wrote: I think I said it wrong. I'm not looking to "pre-compile" the dialplan; No, I understood what you meant :-) Looking for more of a syntax checker than anything. Something like that isn't possible? Yes, but it won't look into the data provided in each extension priority, because that is parsed by the apps themselves. In addition, there are various bits of the dialplan load (merging contexts together, etc.) that can't be done without looking at the one that is currently in memory, or it would miss potential errors. Sorta like "extensions reload [check]" or "extensions check|reload" I agree it would be handy to have, but I think the implementation is non-trivial at best. ___ 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] A good SIP receptionist phone
Rumour has is that Polycom will be releasing a reception console... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Brown Sent: Sunday, 1 May 2005 5:08 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] A good SIP receptionist phone I have a problem. The average person is too freaking stupid to use a VOIP phone. My experience has so far been that if it doesn't have 20 buttons with little red LED's on it, the user cannot comprehend call parking, attended transfer, blind transfer, DND, and navigating through a voicemail menu. I need a good receptionist phone that works with Asterisk. It basically needs to act like an avaya partner phone, I don't need 20 buttons with little red LED's...what I do need is for the phone to register multiple extensions to my asterisk server and act like each SIP extension is a line, so if the idiot receptionist has a call ringing in on line 1, she can pick it up, look at the buttons, see a call ringing in on line 2 (and the phone ringer rings), put call 1 on hold without hanging the caller up, and hit the little "I am an idiot and need a line 2 button" to pick up line 2, so on and so forth. I love VOIP systems and all the functionality they bring and features I get. Unfortunately, the average person in this country anymore is apparently completely stupid and cannot understand how to juggle calls without hanging up on people. So seriously does anyone have a recommendation for a good receptionist phone? I tried the Snom today and I can't get the programmable buttons to do this, even by following the manual. So please, any suggestions would be great, before I get fired at my dayjob for everyone else's idiocy. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ 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] Paging and intercom
I was under the impression that the intercom function on the Snom phones only worked if you did some hacking to asterisk...is that still the case? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Saturday, 30 April 2005 7:33 AM To: Jacob Cazzell; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Paging and intercom Polycom phones and Snom phones supoprt paging. As far as your Overhead paging all you need is an FXO port on your system. The * system will work perfectly with this. Even allowing the zones to be set from the dialplan so your users won't need to learn any new 'paging codes' Email me off -list of you need some help. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacob Cazzell Sent: Friday, April 29, 2005 5:21 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Paging and intercom Hello all, We are considering implementing a new system based on Asterisk on the back end. I am very intrigued by the IP phones, but I have two questions regarding paging and intercom functions. I know that * supports these functions, but I'm not sure I fully understand how. On our existing phone system, if you dial an extention the other end will beep and then setup an intercom channel that's hands free for the called station. I'm not sure how this would be duplicated in *, or is it more of a function of the phone we use? We also have an overhead paging system, our current system is tied into a Valcom 3-zone paging system. Would * support this paging system? How do you get a connection to it, an analog port? These are probably my two biggest hurdles to overcome and I need some pointers on how to implement or where to research my options. Thanks! Jacob ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ 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] Pre-Parse Extensions.conf?
> Matthew Boehm wrote: >> Is there a way to pre-parse your extensions.conf so you can check for errors >> before making it live? > > No, there is not. And any pre-parser that existed today would be > incomplete anyway, because most of the dialplan is not actually parsed > until the applications are called (i.e. it's not parsed a load time, but > at run time). > > There have been discussions about improving the dialplan/application > interfaces in a way that would allow for load-time parsing, but there > hasn't been any movement in that direction yet. If that does happen, > you'd at least be able to load your dialplan in a separate "testing" > instance of Asterisk to see if it parses properly. I think I said it wrong. I'm not looking to "pre-compile" the dialplan; Looking for more of a syntax checker than anything. Something like that isn't possible? It happens when you call "extensions reload" so I figured the function(s) that read and parse the extensions.conf could accept an extra variable to where when the final function that actually loads it into memory is skipped. Sorta like "extensions reload [check]" or "extensions check|reload" -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX channels do not disconnect
Hey I'm experiencing this strange thing. My setup is : PSTN => Asterisk =IAX=> Asterisk => PSTN The thing is when I dial from the PSTN to the asterisk server and the latter has called the PSTN number on the other asterisk server, even if I hangup both the phnes on the PSTN, the ASterisk servers still are talking to each other over IAX. The IAX channel is not hanging up when a PSTN phone hangs up! Please help Regards Riz ___ 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] Polycom IP500 - Phone TIme
It now works - but only in the latest (1.5+) firmware releases. Later, PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Scobie Sent: Friday, 29 April 2005 6:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 - Phone TIme Paul Hales wrote: > And my dreamthat one day Polycom phones will support Australian Daylight > savings... > > But it's only a dream. Unless I am missing something, you don't need to dream about it - set it in ipmid.cfg. Look at the Sip Admim PDF for an explanation of: tcpIpApp.sntp.daylightSavings.enable="1" tcpIpApp.sntp.daylightSavings.fixedDayEnable="0" tcpIpApp.sntp.daylightSavings.start.month="4" tcpIpApp.sntp.daylightSavings.start.date="1" tcpIpApp.sntp.daylightSavings.start.time="2" tcpIpApp.sntp.daylightSavings.start.dayOfWeek="1" tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth="0" tcpIpApp.sntp.daylightSavings.stop.month="10" tcpIpApp.sntp.daylightSavings.stop.date="1" tcpIpApp.sntp.daylightSavings.stop.time="2" tcpIpApp.sntp.daylightSavings.stop.dayOfWeek="1" tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth="1" Regards, Richard ___ 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 CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ 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] Latest CVS Head Nukes Server
Has anyone experienced problems with recent CVS HEAD (as of 30th April) version of * completely crashing the PC on shutdown? (I can't see the console, because the server is located in remote data centre). The problem doesn't appear to happen all the time. Only when * has been running for a while. ___ 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] Centos - Hylafax Install
Does this sound like the centos mailing list. Check irc, #centos On Sun, 1 May 2005, mr. barker wrote: Has anyone tried to install Hylafax on Centos ? If so is there an rpm .. or what was your compiling procedure ? Thanks in return ___ 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] Centos - Hylafax Install
Has anyone tried to install Hylafax on Centos ? If so is there an rpm .. or what was your compiling procedure ? Thanks in return ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Dutch SIP or IAX numbers
> Message: 1 > Date: Sun, 1 May 2005 19:01:24 +0200 > From: Michiel van Baak <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Dutch SIP or IAX numbers > To: asterisk-users@lists.digium.com > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=us-ascii > > On 12:23, Sun 01 May 05, Asterisk wrote: > > How knows where I can get a Dutchphone number for asterisk? > > > > Pilmo is not delivering one for home use. www.voipgate.nl Not using it, but offers IAX2. Wessel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.0 - Release Date: 29-04-05 ___ 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] Web interface Suggestions
> I think you will find AMP is about to implement a multi tenant solution. But does AMP deal with realtime? or just flat files the data for which is held in a db? >> >> Open Source project I assume. I am interested in this project do you Only open source. >> have a webpage about it? You can find the current version at https://www.dalmany.co.uk/asterisk/index.html I am open to suggestions and requests. Pages waiting incorporation include voicemail, sip users and sip peers. This only deals with Realtime, it does not replicate AMP with a db and flatfiles. It does not modify any flatfiles, only the realtime database so one has to know about realtime and how it works to get the full benefit. I am in the throws of moving house which is preventing me from developing it as quickly as I would like. >> >> Thanks, >> _ >> /-\ ndrew >> >> On 4/28/05, G.Marshall <[EMAIL PROTECTED]> wrote: >> > > Has anyone come across any software that can control >> adding/editing >> > > SIP extension properties and perhaps dial plan properties on a > context >> > > basis. What I mean is I would like it so an admin user from > Company A >> > > can manipulate >> > > properties for extensions in his context but not in another > Companies. >> I >> > > know AMP does something similar >> > > to this but from what I understand it does not allow for different >> users >> > > at different companies to control >> > > only things that pertain to them. >> > In my spare time, I am developing a php webfrontend to realtime > asterisk >> > database which modifies dialplan, users etc. Should not be too >> difficult >> > to add a login facility which means the user can see their own > context >> > only. >> > >> > Regards, >> > >> > Spencer >> > --- >> > https://www.dalmany.co.uk/dundi/dundi.php >> > https://www.dalmany.co.uk/asterisk/index.php > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 401 Unauthorized.
I'm having ongoing registration fits with some SPA-2000's. Right now I have one which, based on the debugging output repeatedly fails with "401 unauthorized": - <-- SIP read from 206.127.114.240:5060: REGISTER sip:voip-proxy.mt.net SIP/2.0 Via: SIP/2.0/UDP 206.127.114.240:50291;branch=z9hG4bK-f4e3f6bc From: MIC Sipura User ;tag=abf0b0fbea0a3e68o0 To: MIC Sipura User Call-ID: [EMAIL PROTECTED] CSeq: 52 REGISTER Max-Forwards: 70 Contact: MIC Sipura User ;expires=60 User-Agent: Sipura/Sipura/SPA2000-2.0.13(g) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura --- (12 headers 0 lines)--- Using latest request as basis request Sending to 206.127.114.240 : 50291 (NAT) Transmitting (NAT) to 206.127.114.240:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 206.127.114.240:50291;branch=z9hG4bK-f4e3f6bc;received=206.127.114.240;rport=5060 From: MIC Sipura User ;tag=abf0b0fbea0a3e68o0 To: MIC Sipura User Call-ID: [EMAIL PROTECTED] CSeq: 52 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 --- Transmitting (NAT) to 206.127.114.240:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 206.127.114.240:50291;branch=z9hG4bK-f4e3f6bc;received=206.127.114.240;rport=5060 From: MIC Sipura User ;tag=abf0b0fbea0a3e68o0 To: MIC Sipura User ;tag=as3bb6fc64 Call-ID: [EMAIL PROTECTED] CSeq: 52 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: WWW-Authenticate: Digest realm="asterisk", nonce="36609d12" Content-Length: 0 - This box will register and then be ok for several days and then will go into this mode. I've seen other sipuras doing something similar but with 407. In fact, calls were made with this box yesterday with absolutely no configuration changes on either end, until I started to try to figure out why it isn't registering. With only 5-6 SPA-2000's in test and several of them acting flaky registration-wise I'm feeling that I'm missing something which causes this flakiness. Both the SPA and asterisk have been rebooted. Asterisk has actually been updated to the latest CVS version today in case there was an already-in-cvs fix. The config I have in asterisk for this sipura box is: [A0974L1] type=friend host=dynamic context=cosinternational secret= callerid="MIC Sipura User" <406###> dtmfmode=rfc2833 reinvite=no canreinvite=no nat=yes qualify=no Ideas? Other places I should look? -forrest ___ 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] Execute commands after phone hangs up
On 00:17, Mon 02 May 05, Shady wrote: > The easiest way is to use the "g" parameter of the Dial application. Then > in the next priority you may use the System application to execute an > external mail command. > > > - Original Message - > From: "Chuck Smith" <[EMAIL PROTECTED]> > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > > Sent: Monday, May 02, 2005 12:11 AM > Subject: [Asterisk-Users] Execute commands after phone hangs up > > > >Here is a good one. > > > > > >If I wanted to record a call when it comes into a particular extension, > >then > >e-mail the wav file to a particular email address what would I do? I know > >how to initiate the recording but I don't know how to initiate the e-mail > >after the call is complete. I can initiate a call before the call is > >complete but not after. > > > >Thanks in advance. > > > >Chuck > > Or you can use exten => h,1,System in that particular extension -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence." ___ 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] Execute commands after phone hangs up
The easiest way is to use the "g" parameter of the Dial application. Then in the next priority you may use the System application to execute an external mail command. - Original Message - From: "Chuck Smith" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Monday, May 02, 2005 12:11 AM Subject: [Asterisk-Users] Execute commands after phone hangs up Here is a good one. If I wanted to record a call when it comes into a particular extension, then e-mail the wav file to a particular email address what would I do? I know how to initiate the recording but I don't know how to initiate the e-mail after the call is complete. I can initiate a call before the call is complete but not after. Thanks in advance. Chuck ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Execute commands after phone hangs up
Here is a good one. If I wanted to record a call when it comes into a particular extension, then e-mail the wav file to a particular email address what would I do? I know how to initiate the recording but I don't know how to initiate the e-mail after the call is complete. I can initiate a call before the call is complete but not after. Thanks in advance. Chuck ___ 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] 4 - 8 port w/QOS switch for Asterisk
Yes, I've seen one and comparing it to: The MIL-S8001TG a Layer 2 Gigabit Ethernet Smart switch that features eight 10/100/1000 Copper ports plus a Mini-GBIC (SFP) interface slot. I don't need a router, as I'm using Freesco with print-server and port knocking configuration. Though Freesco doesn't do QOS. #Joseph On Sun, 2005-05-01 at 12:43 -0700, Max W Blackmer Jr wrote: > Linksys has a low end router with an 8 port switch that does QoS model > BEFSR81. It can be gotten for under $100 USD. For more information > http://www.linksys.com/products/product.asp?prid=604&scid=29 > > Max W. Blackmer, Jr. > > > Original Message > > Subject: [Asterisk-Users] 4 - 8 port w/QOS switch for Asterisk > > From: Joseph <[EMAIL PROTECTED]> > > Date: Sun, May 01, 2005 1:05 pm > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > > > Can anybody recommend Switch 4 to 8 ports with QOS for Asterisk? > > > > -- > > #Joseph > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Recording calls
Wouldn't that example kill the call after 15 seconds? I use option 'b' in Monitor() also. Seems to cut down on recordings where you hear lots of ringing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: Sunday, May 01, 2005 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Recording calls On Sun, 2005-05-01 at 15:18 -0300, Jozeph Brasil wrote: > Hi guys, > > I need to record all incoming calls. Anyone know how to do this? > > Thanks, > Jozeph Very easy, take a look: exten => 718,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) exten => 718,2,Monitor(wav,${CALLFILENAME},m) exten => 718,3,AbsoluteTimeout(15); retun control to T exten => 718,4,Dial(${phone1},20,rw) -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pre-Parse Extensions.conf?
Matthew Boehm wrote: Is there a way to pre-parse your extensions.conf so you can check for errors before making it live? No, there is not. And any pre-parser that existed today would be incomplete anyway, because most of the dialplan is not actually parsed until the applications are called (i.e. it's not parsed a load time, but at run time). There have been discussions about improving the dialplan/application interfaces in a way that would allow for load-time parsing, but there hasn't been any movement in that direction yet. If that does happen, you'd at least be able to load your dialplan in a separate "testing" instance of Asterisk to see if it parses properly. ___ 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] Problems in new implemenation....
I have recently implemented a SIP VoIP implementation using Asterisk. I can go through and place a call to a particular number from the PSTN, the phone rings, but I am not getting the ring response back to the calling party. I am not sure as to where this problem is coming from, but I know it stopped working once I added the configurations dial-peer voice 82010151 pots incoming called-number 2010151 direct-inward-dial forward-digits all ! dial-peer voice 2010151 voip destination-pattern 2010151 session protocol sipv2 session target ipv4:XXX.XXX.XXX.XXX session transport udp incoming called-number 2010151 dtmf-relay sip-notify rtp-nte codec g711ulaw ! dial-peer voice 82010152 pots incoming called-number 2010152 direct-inward-dial forward-digits all ! dial-peer voice 2010152 voip destination-pattern 2010152 session protocol sipv2 session target ipv4: XXX.XXX.XXX.XXX session transport udp incoming called-number 2010152 dtmf-relay sip-notify rtp-nte codec g711ulaw ! ! sip-ua max-forwards 15 retry invite 10 timers trying 1000 timers expires 30 sip-server ipv4: XXX.XXX.XXX.XXX no transport tcp ! I also have a Cisco Call Manager Express sending and receiving calls to and from this same equipment without the problem existing. I am sure that this problem is something with the way that I have the SIP commands configured on this AS5400, but I just do not know enough to fix it. Thanks for your thoughts. Stephen <>___ 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] 4 - 8 port w/QOS switch for Asterisk
Linksys has a low end router with an 8 port switch that does QoS model BEFSR81. It can be gotten for under $100 USD. For more information http://www.linksys.com/products/product.asp?prid=604&scid=29 Max W. Blackmer, Jr. > Original Message > Subject: [Asterisk-Users] 4 - 8 port w/QOS switch for Asterisk > From: Joseph <[EMAIL PROTECTED]> > Date: Sun, May 01, 2005 1:05 pm > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Can anybody recommend Switch 4 to 8 ports with QOS for Asterisk? > > -- > #Joseph > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queues configuration
Worth taking a look..thx! |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Kevin P. Fleming |Sent: Domingo, 01 de Mayo de 2005 02:08 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Queues configuration | |Anton Krall wrote: |> I have my agents defined in agents.conf.. Damn.. I normally use |> agentcallbacklogin.. So how can I use agentcallbacklogin and |addqueuemember? | |AddQueueMember does pretty much the same thing as |AgentCallbackLogin, it causes the queue to dial the agent when |a call is being delivered to them. It just doesn't use any of |the chan_agent infrastructure, so you'd have to build your own |agent code/password entry and validation before calling |AddQueueMember. There is an example on the wiki of doing this. |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Programing a call forward feature to cel phones
My other question is.. Why does that sound also happen sometimes while in a call with a pstn number? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Tim Connolly |Sent: Domingo, 01 de Mayo de 2005 02:14 p.m. |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Programing a call forward |feature to cel phones | |Maybe turn echotraining off altogether.. I wonder if the cell |company is also doing some line conditioning that is killing |the call quality after the training (at both ends) stops. | |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Anton Krall |Sent: Sunday, May 01, 2005 2:05 PM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Programing a call forward |feature to cel phones | |I tried eventhough the call are not been "bridge" like when |Asterisk bridges |2 sip calls and steps out of the way. | |I tried using this: | |echocancel=yes |echocancelwhenbridged=yes |echotraining=yes |echotraining=800 | |No luck either :( | |Any ideas? | ||-Original Message- ||From: [EMAIL PROTECTED] ||[mailto:[EMAIL PROTECTED] On Behalf Of Michiel ||van Baak ||Sent: Domingo, 01 de Mayo de 2005 02:52 a.m. ||To: asterisk-users@lists.digium.com ||Subject: Re: [Asterisk-Users] Programing a call forward |feature to cel ||phones || ||On 20:28, Sat 30 Apr 05, Anton Krall wrote: ||> :) ||> No problem dialing another cell phone from asterisk or |incoming from ||> cel phone, etc. ||> ||> Console says nothing. ||> ||> The forwarded call is been directed using zap (x100) ||> ||> So nothing looks wrong... But still...cant figure out why |forwarding ||> the call to a cel phone via zap gets those weird sounds after 2 ||> seconds of talking and why this happens just when ||redirecting to a cel ||> phone. Seems that if you redirect to a land line is ok. ||> ||> Also, sometimes, when in a call, any call (cel, land line, etc) ||> sometimes a weird sound much like the one I mentioned kicks in the ||> call and I cant get the caller because of the sound and he ||cant listen ||> to me, so I need to hit flash and then flash again and the ||call continues without the sounds... ||> Anybody seen that before? Could it be asterisk or the x100? Maybe ||> worth mentioning, that I use Monitor to records all calls... ||Could that be it ? ||> || ||I saw in another post you have echocancelwhenbridged=off Did |you try to ||turn this on and see if the problem is gone ? ||-- ||Michiel van Baak ||http://lunteren.vanbaak.info ||[EMAIL PROTECTED] ||GnuPG key: |http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D || ||"Two of the most famous products of Berkeley are LSD and BSD. ||I don't think that this is a coincidence." || ||___ ||Asterisk-Users mailing list ||Asterisk-Users@lists.digium.com ||http://lists.digium.com/mailman/listinfo/asterisk-users ||To UNSUBSCRIBE or update options visit: || http://lists.digium.com/mailman/listinfo/asterisk-users || | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P does not detect hangup on UK BT analogue line
Dear Collective ... I know that this problem crops up again and again, but I've yet to find something that works for me. I've completely exhausted Google. I have a TDM400P card with a single FXO module connected to a standard analogue BT telephone line. The card works fine, there are no IRQ issues or crackling or echoing or any of that rubbish. However, the card and/or Asterisk fail to detect when the remote party have hungup. This isn't a problem if it's a real telephone conversation with a user of the system as in such cases Asterisk detects the hangup from the local end and terminates the channel, but if someone calls in and gets to an un-manned service such as an IVR menu or voicemail and then hangs up, the channel remains open and you get the "2 minutes of continous tone" voicemail problem. I have correctly modprobed wcfxs with "UK" as the opermode (dmesg confirms): Zapata Telephony Interface Registered on major 196 PCI: setting IRQ 11 as level-triggered PCI: Assigned IRQ 11 for device :00:08.0 Freshmaker version: 71 Freshmaker passed register test Module 0: Not installed Module 1: Not installed Module 2: Not installed Module 3: Installed -- AUTO FXO (UK mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 4 (United Kingdom) This is my /etc/zaptel.conf: loadzone=uk defaultzone=uk fxsks=4 This is my /etc/asterisk/zapata.conf: [channels] language=en context=default signalling=fxs_ks usecallerid=yes cidsignalling=v23 cidstart=polarity hidecallerid=no restrictcid=no callwaiting=no echocancel=yes busydetect=yes callprogress=yes hanguponpolarityswitch=yes echocancel=yes rxgain=4.5 txgain=4.5 immediate=no context=incoming channel => 4 All the polarity-based caller-ID stuff works, so I know that the card is capable of detecting polarity switches, but the (ideal looking) "hanguponpolarityswitch" parameter has no effect. What am I missing? Surely this can't be a bug! Thanks Stuart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recording calls
On Sun, 2005-05-01 at 15:18 -0300, Jozeph Brasil wrote: > Hi guys, > > I need to record all incoming calls. Anyone know how to do this? > > Thanks, > Jozeph Very easy, take a look: exten => 718,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) exten => 718,2,Monitor(wav,${CALLFILENAME},m) exten => 718,3,AbsoluteTimeout(15); retun control to T exten => 718,4,Dial(${phone1},20,rw) -- #Joseph ___ 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] Pre-Parse Extensions.conf?
Is there a way to pre-parse your extensions.conf so you can check for errors before making it live? ARA Extensions is a really cool tool and will allow us to let our customers create/manage their own dialplans. It would be nice if when a customer changes their dialplan that it gets parsed and checked for errors before being inserted into the database. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Recording calls
You might check the wiki next time before you ask: http://www.voip-info.org/wiki-Asterisk+cmd+Monitor http://www.voip-info.org/wiki-Asterisk+cmd+record Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jozeph Brasil Sent: Sunday, May 01, 2005 1:19 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Recording calls Hi guys, I need to record all incoming calls. Anyone know how to do this? Thanks, Jozeph ___ 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] Programing a call forward feature to cel phones
Maybe turn echotraining off altogether.. I wonder if the cell company is also doing some line conditioning that is killing the call quality after the training (at both ends) stops. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Sunday, May 01, 2005 2:05 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Programing a call forward feature to cel phones I tried eventhough the call are not been "bridge" like when Asterisk bridges 2 sip calls and steps out of the way. I tried using this: echocancel=yes echocancelwhenbridged=yes echotraining=yes echotraining=800 No luck either :( Any ideas? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Michiel van Baak |Sent: Domingo, 01 de Mayo de 2005 02:52 a.m. |To: asterisk-users@lists.digium.com |Subject: Re: [Asterisk-Users] Programing a call forward |feature to cel phones | |On 20:28, Sat 30 Apr 05, Anton Krall wrote: |> :) |> No problem dialing another cell phone from asterisk or incoming from |> cel phone, etc. |> |> Console says nothing. |> |> The forwarded call is been directed using zap (x100) |> |> So nothing looks wrong... But still...cant figure out why forwarding |> the call to a cel phone via zap gets those weird sounds after 2 |> seconds of talking and why this happens just when |redirecting to a cel |> phone. Seems that if you redirect to a land line is ok. |> |> Also, sometimes, when in a call, any call (cel, land line, etc) |> sometimes a weird sound much like the one I mentioned kicks in the |> call and I cant get the caller because of the sound and he |cant listen |> to me, so I need to hit flash and then flash again and the |call continues without the sounds... |> Anybody seen that before? Could it be asterisk or the x100? Maybe |> worth mentioning, that I use Monitor to records all calls... |Could that be it ? |> | |I saw in another post you have echocancelwhenbridged=off Did |you try to turn this on and see if the problem is gone ? |-- |Michiel van Baak |http://lunteren.vanbaak.info |[EMAIL PROTECTED] |GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D | |"Two of the most famous products of Berkeley are LSD and BSD. |I don't think that this is a coincidence." | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Make Webvmail Error
I did a make webvmail and I get the following error on redhat 9.0 No HTTP directory make: *** [webvmail] Error 1 I have the perl-suidperl rpm installed and apache installed Thanx. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queues configuration
Anton Krall wrote: I have my agents defined in agents.conf.. Damn.. I normally use agentcallbacklogin.. So how can I use agentcallbacklogin and addqueuemember? AddQueueMember does pretty much the same thing as AgentCallbackLogin, it causes the queue to dial the agent when a call is being delivered to them. It just doesn't use any of the chan_agent infrastructure, so you'd have to build your own agent code/password entry and validation before calling AddQueueMember. There is an example on the wiki of doing this. ___ 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] Intel 536EP
On Saturday 30 April 2005 18:09, Jeff wrote: > Will the Intel 536EP function as a FXO? No. Cheers, Gavin. ___ 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] Programing a call forward feature to cel phones
I tried eventhough the call are not been "bridge" like when Asterisk bridges 2 sip calls and steps out of the way. I tried using this: echocancel=yes echocancelwhenbridged=yes echotraining=yes echotraining=800 No luck either :( Any ideas? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Michiel van Baak |Sent: Domingo, 01 de Mayo de 2005 02:52 a.m. |To: asterisk-users@lists.digium.com |Subject: Re: [Asterisk-Users] Programing a call forward |feature to cel phones | |On 20:28, Sat 30 Apr 05, Anton Krall wrote: |> :) |> No problem dialing another cell phone from asterisk or incoming from |> cel phone, etc. |> |> Console says nothing. |> |> The forwarded call is been directed using zap (x100) |> |> So nothing looks wrong... But still...cant figure out why forwarding |> the call to a cel phone via zap gets those weird sounds after 2 |> seconds of talking and why this happens just when |redirecting to a cel |> phone. Seems that if you redirect to a land line is ok. |> |> Also, sometimes, when in a call, any call (cel, land line, etc) |> sometimes a weird sound much like the one I mentioned kicks in the |> call and I cant get the caller because of the sound and he |cant listen |> to me, so I need to hit flash and then flash again and the |call continues without the sounds... |> Anybody seen that before? Could it be asterisk or the x100? Maybe |> worth mentioning, that I use Monitor to records all calls... |Could that be it ? |> | |I saw in another post you have echocancelwhenbridged=off Did |you try to turn this on and see if the problem is gone ? |-- |Michiel van Baak |http://lunteren.vanbaak.info |[EMAIL PROTECTED] |GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D | |"Two of the most famous products of Berkeley are LSD and BSD. |I don't think that this is a coincidence." | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?
Along the same lines, is there some sort of capacity chart that maps capacity based on translations/transcoding? - Daniel On May 1, 2005, at 2:45 PM, Greg Boehnlein wrote: On Sun, 1 May 2005, David John Walsh wrote: what sort of level of PC is required for 300 concurrent calls? Doing what? Ulaw? That could probably be done on a single P4 2.8 Ghz. If you want to transcode from Ulaw to something else, you need to scale the hardware appropriately. Every case is different. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queues configuration
I have my agents defined in agents.conf.. Damn.. I normally use agentcallbacklogin.. So how can I use agentcallbacklogin and addqueuemember? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Kevin P. Fleming |Sent: Domingo, 01 de Mayo de 2005 11:24 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Queues configuration | |Anton Krall wrote: |> Weird.. |> |> I also have joinwhenempty=no and user can still go into the queue |> without any agents logged in. | |Are you using queue members (specified in queues.conf or via |AddQueueMember()), or using agents (specified in agents.conf)? |If the latter, then the "whenempty" functions won't work, |because as far as app_queue is concerned the queue is never |"empty" of members (since it only sees the agent definitions, |not whether they are logged in or not). |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?
On Sun, 1 May 2005, David John Walsh wrote: > what sort of level of PC is required for 300 concurrent calls? Doing what? Ulaw? That could probably be done on a single P4 2.8 Ghz. If you want to transcode from Ulaw to something else, you need to scale the hardware appropriately. Every case is different. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] Audio cut off at beginning of call
When we call out from our Asterisk system we consistenly lose the first roughly 1500 milliseconds of the audio from the destination. This is easiest to demonstrate with a recorded announcement. In other words, "Hello" for example is missing. We are calling over the PSTN via a voice T1 line. We are using the "stable" cvs from about April 1. I searched lists.digium.com but did not find anyone with this problem using the PSTN. Does anyone have any ideas? Gene Naden, MA , MD Programmer Analyst GlobalTeldata II, LLC 4700 N. Ravenswood Chicago, IL 60640 (773) 878-3161 x 223 ___ 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] TDM400P Power Connector
Mojo Jojo wrote: In the meantime can the card run without the power connector if it has only one module on it? Power is required to generate ringing voltage for FXS modules; if you have only FXO modules, power is not required at all. The number of modules is not relevant, as the ringing generator will never draw power from the PCI bus. ___ 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] Recording calls
Hi guys, I need to record all incoming calls. Anyone know how to do this? Thanks, Jozeph ___ 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] 4 - 8 port w/QOS switch for Asterisk
Can anybody recommend Switch 4 to 8 ports with QOS for Asterisk? -- #Joseph ___ 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] [Announce] New chan_sccp release adds support forCisco 7970
Downloaded and did the 'make' Installed seamlessly... However my 7920 now keeps coming back saying can't find call manager 0 I get this in the cli Attempted to check MWI for NULL device == >> Got message AlarmMessage Alarm Message: Severity: 2, 25: Name=SEP000D282E89AA Load=..-(0.0) Last=Initialized [2049/1234] No length in read: Success (errno 0) == Sending Packet Type KeepAliveAckMessage (4 bytes) Any ideas? Thanks - Original Message - From: "Julien Goodwin" <[EMAIL PROTECTED]> To: Cc: Sent: Sunday, May 01, 2005 10:27 AM Subject: [Asterisk-Users] [Announce] New chan_sccp release adds support forCisco 7970 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Pattern Matching
Hmm.. The only reason it *should* do that is if it runs out of priorities on the more significant match, it will then drop back to the next priority on the next less significant match. Send me your real contexts offline, maybe we’re both missing something in the translation to the list. The ‘incoming’ extensions are 100% match, right??? There’s no 9 or 1 prepended on the inbound call? The reason I ask is usually Vonage and BV send you 1+.. rather than just the 10 digit dnis. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo Jojo Sent: Sunday, May 01, 2005 11:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Pattern Matching I do this already with outgoing calls and it works fine as long as I am only using the Dial command. Where I am running into trouble is when doing something like I have created below. I know the syntax is not 100% correct, just using it as a quicky example. What happens here is if the DNIS matches one of the first two exact numbers, it plays the background, sets the timeouts then goes on and plays the sound in the include and hangs up. What I want it to do is execute the stuff in the include ONLY if none of the exact matches ocurr. I would think this is the way it should work but I can't seem to make it happen. [incoming] Exten => 2145550001,1,Answer Exten => 2145550001,2,Wait(1) Exten => 2145550001,3,Background(MyGreeting) Exten => 2145550001,4,Timeout(30) Exten => 2145550001,5,DigitTimeout(3) Exten => 2145550002,1,Answer Exten => 2145550002,2,Wait(1) Exten => 2145550002,3,Background(MyGreeting) Exten => 2145550002,4,Timeout(30) Exten => 2145550002,5,DigitTimeout(3) Include => Pattern-Include [Pattern-Include] Exten => _8XXNXX,1,Answer Exten => _8XXNXX,2,Wait(1) Exten => _8XXNXX,3,Playback(NumNotConfigured) Exten => _8XXNXX,4,Hangup Private Label Wholesale Internet Access! http://www.YourOwnISP.com - Original Message - From: Tim Connolly To: 'Asterisk Users Mailing List - Non-Commercial Discussion' ; [EMAIL PROTECTED] Sent: Saturday, April 30, 2005 4:21 PM Subject: RE: [Asterisk-Users] Pattern Matching Like this: [dids] Exten => 2145550001,1,dial(SIP/6001) Exten => 2145550002,1,dial(SIP/6002) Exten => 2145550003,1,dial(SIP/6003) Include => default-did [default-did] Exten => _.,1,dial(SIP/6000) Seems pretty simple. I used this method of least/highest cost routing to choose my LD carrier. Should work the same though. http://www.voip-info.org/tiki-index.php?page=Asterisk%20least%20cost%20routing%20using%20broadvoice -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo Jojo Sent: Saturday, April 30, 2005 3:08 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Pattern Matching Not sure what you mean exactly... Can you give me a hint? Private Label Wholesale Internet Access! http://www.YourOwnISP.com - Original Message - From: "Michael D Schelin" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, April 29, 2005 10:10 PM Subject: Re: [Asterisk-Users] Pattern Matching > Hey Mojo, I'm thinking you might try using priorty 's to set some kind > routing. just a thought.. > > > > Mojo Jojo wrote: > >> We recently had our PRI installed, we currently have 100 toll-free's >> pointing to it. >> >> I have almost everything working great but.. >> >> I have setup the first few numbers we want to use coming in from the PRI >> and they work great, but.. >> >> What I want to do is setup an extension with pattern matching to answer >> for any numbers called that are pointed to our system and PRI but not yet >> in use/configured. >> >> I have been successful at setting up pattern matching as a catch all for >> 98 or so numbers not in use yet and I have been successful setting up the >> 2 numbers I want to make use of for now. >> >> Problem is, I can't use both at the same time! >> >> If I turn on the pattern matching then my greeting plays for the >> configured number, then the message plays for the invalid number >> (basically executing the extension with the pattern matching). >> >> I have read about sorting with pattern matching by using an include, I >> did this but it's not really helping. >> >> I have set a response timeout after the first extension plays it's >> greeting, I would think it should wait until it times out but it doesn't, >> it just immediately moves to the pattern matched extension. >> >> I must be missing something big here.. >> >> Any help is appreciated.. >> >> >> -- >> Private Label Wholesale Internet Access! >> http://www.YourOwnI
[Asterisk-Users] TDM400P Power Connector
I have a TDM400P I am trying to install but I need a power connector extender to be able to get power into the card. In the meantime can the card run without the power connector if it has only one module on it? Thanks! Private Label Wholesale Internet Access! http://www.YourOwnISP.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: RE : [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?
what sort of level of PC is required for 300 concurrent calls? Regards David On 5/1/05, Hakem Taourchi <[EMAIL PROTECTED]> wrote: > > > > Can this Dell run 90 calls simultaneously ? Or need a higher Dell machine? > > > > > > -Message d'origine- > De : [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] De la part > de Ariel Batista > Envoyé : samedi 23 avril 2005 1:27 > À : 'Ben Hencke'; 'Asterisk Users Mailing List - Non-Commercial Discussion' > Objet : RE: [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P? > > > > > I just setup a SC420 with two TDMO4b cards in it and it works just fine. No > problems what so ever with it so far. > > > > > > > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf > Of Ben Hencke > Sent: Friday, April 22, 2005 6:42 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P? > > > > I have head that the SC prefixed Dells are not good to use with Digium > hardware. Considering how picky my TE405P cards were in other low end Dell > servers, I would suggest using an 1850 instead. > > OTOH, if it does work, please let me know :-) > If you go to small biz, you can get the SC1425 trimmed down with dual > 2.8hgz for under $1k > - Ben > > > On 4/22/05, Greg Boehnlein <[EMAIL PROTECTED]> wrote: > > Hello, > I've been asked to build a couple of Gateway servers for a client > w/ TE405P hardware, and have been looking around at various 1U options. > I've been looking at SuperMicro and Tyan barbones boxes as possible > platforms, but then was directed to Dell's SC1425 by a friend. Short > story, is that you can purchase a 2x3.0Ghz/1GigDDR400/1xSATA box in a 1U > form factor for $1,498.00. This seems almost too good to be true, so I'm > asking if anyone has had any experience with this box? > > I'm not up on my PCI terminology, but as I understand it, the TE405P can > only be used in a 32 bit 33Mhz slot at 5.0 Volts. This SC1425 lists a > "1x 64-bit/1xxMHz PCI-X slot" under it's expandability information. I'd > venture to guess this is probably NOT going to work with a TE405P. > > That being said, if it works, great. If not, what 1U boxes are people > using IN PRODUCTION w/ TE405P cards? > > -- > Vice President of N2Net, a New Age Consulting Service, Inc. Company > http://www.n2net.net Where everything clicks into place! > KP-216-121-ST > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dutch SIP or IAX numbers
On Sun, 2005-05-01 at 12:23 -0300, Asterisk wrote: > How knows where I can get a Dutchphone number for asterisk? http://www.dis-telecom.nl/v2/dienst.php?id=65 ___ 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] zaptel.conf multiple devices
Hi there my zaptel hardware is giving errors while loading but they seem to load just fine. the lights wil work and my wctdm card is also workin and the isdn works to But when I stop asterisk I have to reload al cards again is this normal? This is my zaptel.conf is there no way to group these because my te110p is giving an error that it can’t find channel 35 but 35 belongs to my wctdm. Maybe my zaptel.conf is not that good, I can’t find any documentation on multiple cards in one system Thanks ZT_SPANCONFIG failed on span 2: No such device or address (6) make: *** [loadlinux26] Error 1 ZT_CHANCONFIG failed on channel 35: No such device or address (6) FATAL: Error running install command for wcte11xp [EMAIL PROTECTED] src]# loadzone=nl defaultzone=nl span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=1,1,0,ccs,hdb3 bchan=4-18,20-34 # set this to 1-15,17-31 for E1 dchan=19 # set this to 16 for E1 fxoks=35-36 ___ 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] Dutch SIP or IAX numbers
On 12:23, Sun 01 May 05, Asterisk wrote: > How knows where I can get a Dutchphone number for asterisk? > > Pilmo is not delivering one for home use. Three I use: http://www.speakup.nl (for now only busines accounts) http://www.talkin2ya.nl (both prepaid and postpaid) http://www.dutchphone.nl (only postpaid) -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence." ___ 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] Asterisk 1.0.6 stable IAX2 Firefly supervised call transfer?
Hi I see various discussions on this but cannot get it to work, and is not clear that anyone resolved this. This seems pretty fundamental so I am missing something, but I cannot find it anywhere. # does work for blind transfers - no problem. But the various * commands given in features.conf do not. OK, I've picked up that this may not be in the released one but also I've found that chan_iax2.c does talk about attended transfers. Also iax2 debug shows that the * key is being recognised and passed back. Can anyone help on this - IAX2 is so much better than SIP which may not have this problem. We're using Firefly phone - neat, simple, seems reliable. Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP Reject Call Option
At 11:27 AM 5/1/2005, you wrote: Asterisk wrote: Is there anyway of having a "Reject Call" button appear when there is an incoming call. Sometimes I am wating for a call, but one from another person comes through - I would like to press a button and send them straight to voicemail. You can press the "EndCall" button while an unanswered call is ringing to achieve the same effect. The only available menu button is "Answer" when an inbound call is ringing on my 7960g. The menu with EndCall does not come up until I answer the call. Tom Sorry for jumping in but I am after the same thing. ___ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP Reject Call Option
Asterisk wrote: Is there anyway of having a "Reject Call" button appear when there is an incoming call. Sometimes I am wating for a call, but one from another person comes through - I would like to press a button and send them straight to voicemail. You can press the "EndCall" button while an unanswered call is ringing to achieve the same effect. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queues configuration
Anton Krall wrote: Weird.. I also have joinwhenempty=no and user can still go into the queue without any agents logged in. Are you using queue members (specified in queues.conf or via AddQueueMember()), or using agents (specified in agents.conf)? If the latter, then the "whenempty" functions won't work, because as far as app_queue is concerned the queue is never "empty" of members (since it only sees the agent definitions, not whether they are logged in or not). ___ 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] Pattern Matching
I do this already with outgoing calls and it works fine as long as I am only using the Dial command. Where I am running into trouble is when doing something like I have created below. I know the syntax is not 100% correct, just using it as a quicky example. What happens here is if the DNIS matches one of the first two exact numbers, it plays the background, sets the timeouts then goes on and plays the sound in the include and hangs up. What I want it to do is execute the stuff in the include ONLY if none of the exact matches ocurr. I would think this is the way it should work but I can't seem to make it happen. [incoming] Exten => 2145550001,1,Answer Exten => 2145550001,2,Wait(1) Exten => 2145550001,3,Background(MyGreeting) Exten => 2145550001,4,Timeout(30) Exten => 2145550001,5,DigitTimeout(3) Exten => 2145550002,1,Answer Exten => 2145550002,2,Wait(1) Exten => 2145550002,3,Background(MyGreeting) Exten => 2145550002,4,Timeout(30) Exten => 2145550002,5,DigitTimeout(3) Include => Pattern-Include [Pattern-Include] Exten => _8XXNXX,1,Answer Exten => _8XXNXX,2,Wait(1) Exten => _8XXNXX,3,Playback(NumNotConfigured) Exten => _8XXNXX,4,Hangup Private Label Wholesale Internet Access!http://www.YourOwnISP.com - Original Message - From: Tim Connolly To: 'Asterisk Users Mailing List - Non-Commercial Discussion' ; [EMAIL PROTECTED] Sent: Saturday, April 30, 2005 4:21 PM Subject: RE: [Asterisk-Users] Pattern Matching Like this: [dids] Exten => 2145550001,1,dial(SIP/6001) Exten => 2145550002,1,dial(SIP/6002) Exten => 2145550003,1,dial(SIP/6003) Include => default-did [default-did] Exten => _.,1,dial(SIP/6000) Seems pretty simple. I used this method of least/highest cost routing to choose my LD carrier. Should work the same though. http://www.voip-info.org/tiki-index.php?page=Asterisk%20least%20cost%20routing%20using%20broadvoice -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo JojoSent: Saturday, April 30, 2005 3:08 PMTo: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Pattern Matching Not sure what you mean exactly... Can you give me a hint? Private Label Wholesale Internet Access! http://www.YourOwnISP.com - Original Message - From: "Michael D Schelin" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, April 29, 2005 10:10 PM Subject: Re: [Asterisk-Users] Pattern Matching > Hey Mojo, I'm thinking you might try using priorty 's to set some kind > routing. just a thought.. > > > > Mojo Jojo wrote: > >> We recently had our PRI installed, we currently have 100 toll-free's >> pointing to it. >> >> I have almost everything working great but.. >> >> I have setup the first few numbers we want to use coming in from the PRI >> and they work great, but.. >> >> What I want to do is setup an extension with pattern matching to answer >> for any numbers called that are pointed to our system and PRI but not yet >> in use/configured. >> >> I have been successful at setting up pattern matching as a catch all for >> 98 or so numbers not in use yet and I have been successful setting up the >> 2 numbers I want to make use of for now. >> >> Problem is, I can't use both at the same time! >> >> If I turn on the pattern matching then my greeting plays for the >> configured number, then the message plays for the invalid number >> (basically executing the extension with the pattern matching). >> >> I have read about sorting with pattern matching by using an include, I >> did this but it's not really helping. >> >> I have set a response timeout after the first extension plays it's >> greeting, I would think it should wait until it times out but it doesn't, >> it just immediately moves to the pattern matched extension. >> >> I must be missing something big here.. >> >> Any help is appreciated.. >> >> >> -- >> Private Label Wholesale Internet Access! >> http://www.YourOwnISP.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 mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >
Re: [Asterisk-Users] TFTP question
can you run tftp manually ? tftp 192.168.2.2 get ata01234567890a *** from the RTFM in the upgrade package.. sata186us version 3.1 This is a newer manual upgrade server software, also previously known as "upgrade.exe" Upgrade for ATA 186/188 NOTE: This software is to be run from dos command line in Windows 9X/ME/NT/2000 Requirements: -) network connection between PC and Cisco ATA 186 -) PC machine running windows O/S To upgrade your box, save this executable and the software image (the software image will have a ".zup" extension) and language image (the language image will have a ".kup" extension) in a directory on your PC. At the DOS prompt of the directory where the files are saved, you will enter the following: sata186us -d1 -any2 WARNING -- When upgrading from software version 1.xx to 2.0, make sure there are entries in the UID0 and UID1 fields, and that you hear a dial tone when you pick up the telephone handset. Failure to do so can result in loss of the MAC address during upgrade. Your screen will prompt you with instructions on how to upgrade the box. You will access the voice prompt of the ATA 186 and enter the following commands: 100#*8000# (to upgrade the ATA 186's software version) 101#*8000# (to upgrade the ATA 186's language file) When upgrading many boxes, you can save time by saving the commands above in your telephone's speed-dial, and using them after accessing the ATA 186's voice menu. Available options: when using this upgrade software: usage: sata186us version 3.1 usage: sata186us {-h[host_ip]} {-p[port]} {-quiet} -h[host_ip] Set host IP to specific IP (in the case where there are more than one IP addresses for the host. Default use 1st IP address obtained by gethostbyname). -p[port] Set server port to specific port (default is 8000, use different port only if you are setting up an IP directed upgrade server other than the default). -quiet quiet mode, send all output to log file named as [port].log (useful when running the upgrade server as a deamon). -any Allow upgrade even if software version is less than or equal to those of client box. -any2 Allow upgrade regardless of software type and version. -d1,-d2,-d3 Set verbose level for debugging. imageFile Image file is file with a '.zup' or '.kup' extension. e.g. sata186us -any -d1 test.zup sata186us -h192.168.2.170 -p8002 -quiet test.zup Below is a sample run of upgrade server in ready mode PC name: mypc PC IP: 192.168.1.10 Microsoft(R) Windows Millennium (C)Copyright Microsoft Corp 1981-1999. C:\>ata186us ata134-elang.kup Using Host: mypc with IP : 192.168.1.10 as upgrade server image found: language 51 -- ata186.itsp2.v1.34 Using dialpad of your telephone (attached to your ATA box), press ATA button to go to main menu, and enter: 101#192*168*1*10*8000# (to upgrade language 51) NOTE: Pressing 123# will announce your code's version number. You can later verify that you have upgraded your ATA box. --- This program runs continuously; Press -c to abort. Upgrade server ready... from: http://voip-info.org/wiki-Asterisk+phone+cisco+ATA18x The latest 2.x release SIP/H.323 firmware: ftp://ftp.rekom.ru/pub/ata18x/ata18x-v2-16-2-030909a-1.zip http://kvin.lv/pub/Cisco/ata18x-v2-16-2-030909a-1.zip 3.1(0) firmware: http://kvin.lv/pub/Cisco/ata_03_01_00_sip_040211_1.zip On Sun, 2005-05-01 at 08:49, Hermann Wecke wrote: I'm trying to use TFTP to update the firmware on a Cisco ATA188 but I'm receiving this error: May 1 06:51:50 mail2 in.tftpd[11499]: connect from 192.168.2.2 May 1 06:51:50 mail2 tftpd[11500]: tftpd: trying to get file: ata01234567890a May 1 06:51:50 mail2 tftpd[11500]: tftpd: serving file from /tftpboot May 1 06:51:50 mail2 in.tftpd[11501]: connect from 192.168.2.2 May 1 06:51:50 mail2 tftpd[11502]: tftpd: trying to get file: ata01234567890a May 1 06:51:50 mail2 tftpd[11502]: tftpd: serving file from /tftpboot May 1 06:51:50 mail2 in.tftpd[11503]: connect from 192.168.2.2 May 1 06:51:50 mail2 tftpd[11504]: tftpd: trying to get file: ata01234567890a May 1 06:51:50 mail2 tftpd[11504]: tftpd: serving file from /tftpboot May 1 06:51:55 mail2 tftpd[11500]: tftpd: read: Connection refused May 1 06:51:55 mail2 tftpd[11502]: tftpd: read: Connection refused May 1 06:51:55 mail2 tftpd[11504]: tftpd: read: Connection refused hosts.allow and hosts.deny are empty, directory /tftpboot and file
Re: [Asterisk-Users] Playback() stops working.
I'm working on configuring asterisk 1.0.7 on Debian Sarge. The servers been tested a bit and seemed to working fine, but for some reason now, when I try and run the Playback() or Background() applications, or even try and goto voicemail asterisk refuses to play any sounds back to me. I have heard from 5 or so people about this problem. I run CVS STABLE (almost the same as 1.0.7) and had none of these issues. I wish I could help you, but I wanted to let you know that this is not a general problem. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] TFTP question
I'm trying to use TFTP to update the firmware on a Cisco ATA188 but I'm receiving this error: May 1 06:51:50 mail2 in.tftpd[11499]: connect from 192.168.2.2 May 1 06:51:50 mail2 tftpd[11500]: tftpd: trying to get file: ata01234567890a May 1 06:51:50 mail2 tftpd[11500]: tftpd: serving file from /tftpboot May 1 06:51:50 mail2 in.tftpd[11501]: connect from 192.168.2.2 May 1 06:51:50 mail2 tftpd[11502]: tftpd: trying to get file: ata01234567890a May 1 06:51:50 mail2 tftpd[11502]: tftpd: serving file from /tftpboot May 1 06:51:50 mail2 in.tftpd[11503]: connect from 192.168.2.2 May 1 06:51:50 mail2 tftpd[11504]: tftpd: trying to get file: ata01234567890a May 1 06:51:50 mail2 tftpd[11504]: tftpd: serving file from /tftpboot May 1 06:51:55 mail2 tftpd[11500]: tftpd: read: Connection refused May 1 06:51:55 mail2 tftpd[11502]: tftpd: read: Connection refused May 1 06:51:55 mail2 tftpd[11504]: tftpd: read: Connection refused hosts.allow and hosts.deny are empty, directory /tftpboot and files are readable by owner/group/others. Running tftpd (0.17-12) on Debian Sarge. Similar error message when running atftp. Ideas? ___ 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] vmail.cgi: -rwsr-sr-x as root *still* won't read the files
On Fri, Apr 29, 2005 at 10:50:42AM -0400, mike castleman wrote: > On Fri, Apr 29, 2005 at 12:23:48AM -0500, Brian Capouch wrote: > > > > Drat. Perl screams bloody murder if you try to just set its SUID bit, > > which of course is dangerous as hell. > > The perl-suid is *not* simply a version of perl with the suid bit set > but rather a helper binary which allows perl to run suid scripts. Try > it. Note that this script does not use any of the standard safety mechanisms perl provides to achive some safety. It does not use -w or -T or strict. Nor is it simple to adapt it to use those. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] vmail.cgi: -rwsr-sr-x as root *still* won't read the files
On Thu, Apr 28, 2005 at 11:43:57PM -0500, Brian Capouch wrote: > I'm running Apache as "nobody." Wondering why the SUID vmail.cgi script > still can't read my files; it comes with the bits set SUID, which I > thought would do the trick. > > It works just fine if I make the files in the maildir world-readable. > > Thanks. No clues in the archives no Wiki that appear germane. apache's suexec will not run suid scripts. It will also not run scripts as root. It has a strict checklist (specified in its docs) that it checks the target script before exeecuting it. If the script fails one of those requirements, you'll see a note in suexec's logs. Linux in general will not run SUID scripts (executables whose magic is '#!') as some race conditions will allow you to abuse this to run arbitrary command as the target user. Anyway, asterisk should not be running as root. It should be running under its own, separate user. That's what the switch -U is for. And now you only have to find a way to run that script as that asterisk user. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] Playback() stops working.
Hello, I'm working on configuring asterisk 1.0.7 on Debian Sarge. The servers been tested a bit and seemed to working fine, but for some reason now, when I try and run the Playback() or Background() applications, or even try and goto voicemail asterisk refuses to play any sounds back to me. How can I start to debug the cause of this? Thanks ~sm ___ 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] Dutch SIP or IAX numbers
How knows where I can get a Dutchphone number for asterisk? Pilmo is not delivering one for home use. Thanks Johannes ___ 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 based fax client software
I know there is a way to receive faxes via asterisk but is there any way to send out faxes using a soft client, something that can be installed on a pc like Winfax that can send out faxes via my asterisk server? I have a packet 8 ata connected via an x100p card and I’d like to be able to fax out from two pc’s on my lan – it seems crazy to have to buy an fxs card for a fax machine when I already run IAX voice clients on both pc’s Cheers, Dean ___ 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] Problem with Sangoma/Adtran 600 installation
x lk x x x Interface Name-> w1g1x x x x Operation Mode-> TDM_VOICE x x x x TDM Voice Span-> 1 x x x x Override Asterisk Echo Enable -> No What should the TDM Voice Span be? Chris Mason www.anguillaguide.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] Problem with Sangoma/Adtran 600 installation
I made some progres on this. The udev issues were causing me problems. I rectified this and was able to build zaptel successfully. Wanrouter start works great. [EMAIL PROTECTED] asterisk]# wanrouter start Starting WAN Router... Loading WAN drivers: wanpipe done. Starting up device: wanpipe1 Configuring interfaces: w1g1 done. The next problem comes here [EMAIL PROTECTED] asterisk]# ztcfg -v Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) 24 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) Chris Mason www.anguillaguide.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] Problem with Sangoma/Adtran 600 installation
On May 1, 2005 08:36 am, Chris Mason (Lists) wrote: > Waiting for zap to come online...Error: missing /dev/zap! Uh... the error seems obvious. Have a good look at the various READMEs in the zaptel source directory. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?
Can this Dell run 90 calls simultaneously ? Or need a higher Dell machine? -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ariel Batista Envoyé : samedi 23 avril 2005 1:27 À : 'Ben Hencke'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : RE: [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P? I just setup a SC420 with two TDMO4b cards in it and it works just fine. No problems what so ever with it so far. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Hencke Sent: Friday, April 22, 2005 6:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P? I have head that the SC prefixed Dells are not good to use with Digium hardware. Considering how picky my TE405P cards were in other low end Dell servers, I would suggest using an 1850 instead. OTOH, if it does work, please let me know :-) If you go to small biz, you can get the SC1425 trimmed down with dual 2.8hgz for under $1k - Ben On 4/22/05, Greg Boehnlein <[EMAIL PROTECTED]> wrote: Hello, I've been asked to build a couple of Gateway servers for a client w/ TE405P hardware, and have been looking around at various 1U options. I've been looking at SuperMicro and Tyan barbones boxes as possible platforms, but then was directed to Dell's SC1425 by a friend. Short story, is that you can purchase a 2x3.0Ghz/1GigDDR400/1xSATA box in a 1U form factor for $1,498.00. This seems almost too good to be true, so I'm asking if anyone has had any experience with this box? I'm not up on my PCI terminology, but as I understand it, the TE405P can only be used in a 32 bit 33Mhz slot at 5.0 Volts. This SC1425 lists a "1x 64-bit/1xxMHz PCI-X slot" under it's expandability information. I'd venture to guess this is probably NOT going to work with a TE405P. That being said, if it works, great. If not, what 1U boxes are people using IN PRODUCTION w/ TE405P cards? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with Sangoma/Adtran 600 installation
> Also, what version of the wanroute driver software are you using? wanpipe-2.3.2-1 > 1. modprobe zaptel Works. [EMAIL PROTECTED] asterisk]# service zaptel start Loading zaptel framework: [ OK ] Waiting for zap to come online...Error: missing /dev/zap! Chris Mason www.anguillaguide.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] Problem with PSTN
> I want to use this to call on to a Telecom line(PSTN) and vice versa. I read > somewhere that we need to use some provider for it like FWD or iconnect, do > we need to use them to make outgoing and incoming calls to PSTN lines or we > can do it without them. Try reading these articles: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html >I don't know what should I put in the .conf files so > that it enables these calls. Download and print the available PDFand read it a few times http://www.asteriskdocs.org All the answers to your current questions (and more) are there. ___ 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] IPS Version 0.112 released
IPSwitchBoard Version 0.112 - 1. may 2005. Manual for IPSwitchBoard is now included in the download for the new version, please let me know if you find errors in the manual. It's now possible to specify a program to launch along with parameters when there's an incoming call for IPS. You can also use the Caller ID Number as parameter. Download: http://ipswitchboard.thorben.dk ___ IPSwitchBoard is a FREE Windows.Net application that will allow you to: Unattended/attended transfers. Park calls and retrieve/forward them again. Organize all your Zap, SIP and IAX extensions (automatically retrieved from Asterisk). Hotel/Call Shop Billing module Monitor all extensions. Monitor all queues. Monitor Agents. Monitor Parked Calls. Dynamically log extensions in and out of queues. Integration with CRM software on the web. Record conversations. Browse Call Records Drop any active call. Set Do Not Disturb on Extensions and give a reason. Speed Dialling. User selectable ring tones for IPSwitchBoard. User selectable button colors. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 one way audio
Duane Cox wrote on Friday, 29 April 2005 10:17 AM: > Do you get 2-way audio that sometimes drops off to 1-way audio then > picks back up as 2-way? (Thats what I see) Not sure if my problem is > a lost packet issue as I am sending IAX off net. My experience has been that there is no two-way audio, except that I am able to send a SINGLE, brief touch tone digit. Let me know if I can provide anything to assist in getting this bug fixed. Sincerely, Trevor Hammonds ___ 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] A good SIP receptionist phone
Hi, Citeren Tim Connolly <[EMAIL PROTECTED]>: > Can you show us an example of using the callerID for this purpose? Simple: exten = 31531234567,1,SetCIDName(My DIDnr 1) exten = 31538901234,1,SetCIDName(My DIDnr 2) exten = _X.,2,Dial(SIP/myphone) This way, the CallerID number is untouched, but the Name is set to your DID. Best regards, Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Announce] New chan_sccp release adds support for Cisco 7970
A new chan_sccp release has just been uploaded which adds support for the cisco 7970 (min version 6 firmware). There is one currently known issue with the 7970 support, MWI doesn't work, and only basic call functions have been tested. I'd like to publicly thank three people who've helped a lot to get this release out: * Jared Mauch for an initial patch * Adam Megacz for a hardware donation of a 7970 to test on * Flexion for supplying some ethereal dumps of a 7970 registering to a callmanager The next aim for chan_sccp is to support the subscribe/notify features in asterisk to give BLF type functionality to the 7970 and 7960+7914. This release has been tested on all the phones I own: * Cisco 12SP+ * Cisco 30VIP * Cisco 7905 * Cisco 7940 * Cisco 7970 If I don't have one I can't test it. I'm especially looking for: * Cisco 7910 * Cisco 7920 * Cisco 7935/36 * And any non-cisco sccp phones (the Kirk IP600 looks very intresting and if there are any Australian resellers of this here give me a call) Please see the chan_sccp donations page if you're able to help (Note that I'm in Australia for those considering hardware donations, but am willing to pay shipping). http://chan-sccp.sourceforge.net/donation.html Thanks, Julien Goodwin chan_sccp project lead pgpKJDYoXVB6b.pgp Description: PGP signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mISDN error while compiling
Hi there all! Does anyone know what this error is??? I am trying to compile the mISDN in kernel 2.6.11.5 I get the same error in kernel 2.6.10.2 Someone?? HELP!!! WARNING: /lib/modules/2.6.11.5/kernel/drivers/isdn/hardware/mISDN/hfcmulti.ko needs unknown symbol pci_find_subsys ___ 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] Programing a call forward feature to cel phones
On 20:28, Sat 30 Apr 05, Anton Krall wrote: > :) > No problem dialing another cell phone from asterisk or incoming from cel > phone, etc. > > Console says nothing. > > The forwarded call is been directed using zap (x100) > > So nothing looks wrong... But still...cant figure out why forwarding the > call to a cel phone via zap gets those weird sounds after 2 seconds of > talking and why this happens just when redirecting to a cel phone. Seems > that if you redirect to a land line is ok. > > Also, sometimes, when in a call, any call (cel, land line, etc) sometimes a > weird sound much like the one I mentioned kicks in the call and I cant get > the caller because of the sound and he cant listen to me, so I need to hit > flash and then flash again and the call continues without the sounds... > Anybody seen that before? Could it be asterisk or the x100? Maybe worth > mentioning, that I use Monitor to records all calls... Could that be it ? > I saw in another post you have echocancelwhenbridged=off Did you try to turn this on and see if the problem is gone ? -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence." ___ 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] Cisco 7960 SIP Reject Call Option
Is there anyway of having a "Reject Call" button appear when there is an incoming call. Sometimes I am wating for a call, but one from another person comes through - I would like to press a button and send them straight to voicemail. Sort of a Dynamic Do Not Disturb ... :) Julian ___ 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] Broadvoice limits???
On Sun, 2005-05-01 at 01:53 -0500, Tim Connolly wrote: > I would know if they real-time charged for that... Although I normally only > have 10-12 calls going, I watch pretty close and dispute any > supposed-to-be-free-but-not calls! The 20-30 loop was this month, and they havent charged, they all show as $0.00 on my detail. As for your 5 outbound limit, I dunno, maybe it was just that time maybe it was something else because all inbound in that loop was stanaphone all outbound 1 single broadvoice account. I dont intentionally abuse that 'feature' and havent done it since their upgrades (which it appears every voip provider did this week) so I dunno. I believe that its a violation, a tech said (which is not binding but ...) that you are supposed to have 1 BV account per *person* using the service, which for a home situation is not relaistic. The tech did use the conditional 'technically' when telling me this. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fwd: Sip calling errors
-- Forwarded message -- From: iMRAN <[EMAIL PROTECTED]> Date: May 1, 2005 12:16 PM Subject: Sip calling errors To: Asterisk Users Mailing List - Non-Commercial Discussion , Alexander Scheerschmidt <[EMAIL PROTECTED]> Hi Pros, I`m new to Asterisk Getting following errors on my * : -- Executing Dial("SIP/1000-ee7c", "SIP/[EMAIL PROTECTED]") in new stack -- Called [EMAIL PROTECTED] Apr 28 21:06:09 WARNING[2268]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/venus-e8ba(2) to SIP/1000-ee7c(256) Apr 28 21:06:09 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot find extension context 'default' Apr 28 21:06:09 NOTICE[2268]: channel.c:1724 ast_set_read_format: Unable to find a path from g729 to gsm Apr 28 21:06:09 NOTICE[2268]: channel.c:1691 ast_set_write_format: Unable to find a path from gsm to g729 -- SIP/venus-e8ba is making progress passing it to SIP/1000-ee7c RFC3389: 1 bytes, level 256... Apr 28 21:06:10 NOTICE[2268]: rtp.c:298 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible Apr 28 21:06:13 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot find extension context 'default' -- SIP/venus-e8ba answered SIP/1000-ee7c -- Attempting native bridge of SIP/1000-ee7c and SIP/venus-e8ba Apr 28 21:06:21 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 25090 (Non-critical Response) Apr 28 21:06:22 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 25090 (Non-critical Response)onse) My SIP.CONF [general] port = 5060 bindaddr = 0.0.0.0 canreinvite=no disallow=all allow=g723.1 allow=alaw allow=ulaw allow=gsm allow=g729 [venus] type=friend context=sip-dial host=2.2.2.2 canreinvite=no qualify=yes disallow=all allow=gsm allow=g729 insecure=very dtmfmode=info tos=0x18 [] type=friend host=dynamic username= secret=imran dtmf=inband context=internal dtmfmode=rfc2833 [1000] type=friend username=1000 ;secret=password1 host=dynamic allow=g729 allow=g723.1 context=internal dtmfmode=rfc2833 = [general] static=yes writeprotect=yes [globals] PHONE1=SIP/ PHONE2=SIP/1000 PHONE3=SIP/1001 [internal] include => local-sip [local-sip] exten => ,1,Dial(${PHONE1},40,t) exten => ,2,Hangup exten => 1000,1,Dial(${PHONE2},40,t) exten => 1000,2,Hangup exten => 1001,1,Dial(${PHONE3},40,t) exten => 1001,2,Hangup exten => _00.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) exten => _00.,2,Hangup Venus is my SIP provider (sorry u might hav guessed already) 1000 and 1001 belongs to my AudioCodec MP108 8 FXS SIP device and is my softphone SJphone, i can dial soft to hard and vise versa, i can call to US number thru my SIP provider using my Sjphone (crapy sound) but when i try to dial from MP108 i get the above errors i mentioned. MP108 have preloaded codec i.e. g729 and g723.1, my provider supports g729 and g723.1 please can anyone help me ? ___ 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 calling errors
Hi Pros, I`m new to Asterisk Getting following errors on my * : -- Executing Dial("SIP/1000-ee7c", "SIP/[EMAIL PROTECTED]") in new stack -- Called [EMAIL PROTECTED] Apr 28 21:06:09 WARNING[2268]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/venus-e8ba(2) to SIP/1000-ee7c(256) Apr 28 21:06:09 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot find extension context 'default' Apr 28 21:06:09 NOTICE[2268]: channel.c:1724 ast_set_read_format: Unable to find a path from g729 to gsm Apr 28 21:06:09 NOTICE[2268]: channel.c:1691 ast_set_write_format: Unable to find a path from gsm to g729 -- SIP/venus-e8ba is making progress passing it to SIP/1000-ee7c RFC3389: 1 bytes, level 256... Apr 28 21:06:10 NOTICE[2268]: rtp.c:298 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible Apr 28 21:06:13 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot find extension context 'default' -- SIP/venus-e8ba answered SIP/1000-ee7c -- Attempting native bridge of SIP/1000-ee7c and SIP/venus-e8ba Apr 28 21:06:21 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 25090 (Non-critical Response) Apr 28 21:06:22 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 25090 (Non-critical Response)onse) My SIP.CONF [general] port = 5060 bindaddr = 0.0.0.0 canreinvite=no disallow=all allow=g723.1 allow=alaw allow=ulaw allow=gsm allow=g729 [venus] type=friend context=sip-dial host=2.2.2.2 canreinvite=no qualify=yes disallow=all allow=gsm allow=g729 insecure=very dtmfmode=info tos=0x18 [] type=friend host=dynamic username= secret=imran dtmf=inband context=internal dtmfmode=rfc2833 [1000] type=friend username=1000 ;secret=password1 host=dynamic allow=g729 allow=g723.1 context=internal dtmfmode=rfc2833 = [general] static=yes writeprotect=yes [globals] PHONE1=SIP/ PHONE2=SIP/1000 PHONE3=SIP/1001 [internal] include => local-sip [local-sip] exten => ,1,Dial(${PHONE1},40,t) exten => ,2,Hangup exten => 1000,1,Dial(${PHONE2},40,t) exten => 1000,2,Hangup exten => 1001,1,Dial(${PHONE3},40,t) exten => 1001,2,Hangup exten => _00.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) exten => _00.,2,Hangup Venus is my SIP provider (sorry u might hav guessed already) 1000 and 1001 belongs to my AudioCodec MP108 8 FXS SIP device and is my softphone SJphone, i can dial soft to hard and vise versa, i can call to US number thru my SIP provider using my Sjphone (crapy sound) but when i try to dial from MP108 i get the above errors i mentioned. MP108 have preloaded codec i.e. g729 and g723.1, my provider supports g729 and g723.1 please can anyone help me ? ___ 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